[asterisk-users] DIDForSale July Special (No Activation on new DID Purchases)
*All, To meet the target for the month, we are running a special promotion. $5 activation fee waived for all new DID purchases.* Buy DIDs from DIDForSale http://www.didforsale.com/ today and *your $5 activation fees will be WAIVED* for all the DIDs purchased before July 20 2009. There is no limit on the number of DIDs you can buy and the offer is valid for all customers on new purchases only. We have inbound DIDs in 2 different configurations. 1) DID with unmetered inbound and 20 channels ($8.99 per DID). Additional channels can be purchased at $1 per additional channel. 2) DID with metered inbound are for $1 per DID and $0.004 (0.4 cents) per minute for all incoming calls. What our customer are saying about us, Please click on the link, http://www.didforsale.com/blog/?p=103 Thank you, www.didforsale.com ___ -- 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] DIDForSale July Special (No Activation on new DID Purchases)
Sincere Apologies-- Send the mail to wrong list, Meant to send to asterisk-biz list. -J On Wed, Jul 8, 2009 at 11:35 PM, Jai Rangi jpra...@gmail.com wrote: *All, To meet the target for the month, we are running a special promotion. $5 activation fee waived for all new DID purchases.* Buy DIDs from DIDForSale http://www.didforsale.com/ today and *your $5 activation fees will be WAIVED* for all the DIDs purchased before July 20 2009. There is no limit on the number of DIDs you can buy and the offer is valid for all customers on new purchases only. We have inbound DIDs in 2 different configurations. 1) DID with unmetered inbound and 20 channels ($8.99 per DID). Additional channels can be purchased at $1 per additional channel. 2) DID with metered inbound are for $1 per DID and $0.004 (0.4 cents) per minute for all incoming calls. What our customer are saying about us, Please click on the link, http://www.didforsale.com/blog/?p=103 Thank you, www.didforsale.com ___ -- 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] Anonymous Connection form IP to use specific Context
If you create a peer definition and put the host address in it and the context you want it to go to you should be fine Cheers Duncan David Klaverstyn wrote: Hi All, I never saw a reply to this question. Is anyone able to assist? Regards David. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Klaverstyn *Sent:* Friday, 19 June 2009 2:28 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Anonymous Connection form IP to use specific Context Hi All, How can I force an anonymous SIP connection from a certain IP address to use a specific context rather than the default one defined in sip.conf. I am using Asterisk 1.6.0.9 Regards *David Klaverstyn* ___ -- 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] CIDlookup
Hi List I've a CID lookup hooked onto an inbound route (i m using trixbox) ...it runs well but it returns the value as CIDNAMECIDNUMBER ... if i just want to display the CIDNAME [leaving the quotes and CIDNUMBER] .. how can i do it ? do i have to edit some macro in extensions.conf ? rgds Sriram___ -- 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] OT - How to indent AEL file
Hi,
As my extensions.ael is becoming quite long (3000 lines), I'm wondering if
existing indentation tools
such as vim, indent, ... could improve its formatting (split long lines into
several ones, align {}, ..)
Has anyone tried ?
Regards
___
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Re: [asterisk-users] Queue autopause
On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co wrote: Christian Gansberger escribió: Hi all! I want to autopause my queue member when they are not answering within 20 seconds, and the autopause should affect all queues they are member of, not only the queue where the call was not answered. Is there a way to do that? The members gets dynamically added. I'm using asterisk 1.4.21.2. ___ -- 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 Why would you want to do that? The purpose of the autopause is to discard the absent agent that is not responding to the calls to not try it anymore until it gets unpaused by a supervisor or someone else, and therefore the pause is made to all queues the agent is member of. Why pause it on only one queue, letting it ring on other queues? Aside from the purpose you have on this, I think you would need to modify the app_queue.c code to make the parameter configurable inside each queue definition and not on the general section of queues.conf. Then you would need to modify the logic to handle the autopause configured for each queue. This is a general idea as I didn't take a deep look of app_queue.c to see how it works exactly. Any other solution without changing asterisk code would imply a external application that monitors the queues and makes the custom autopause you need. Just my two cents... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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 To make things clearer: I want the queue member is autopaused on all queues. As a matter of fact in asterisk (vers. 1.4.24.1) the queue member is only paused on one Queue. I tried setting autopause=yes in general context, which doesn't do anything. So i set autopause=yes in every Queue definition, which is working, but only on that queue. I don't use the agents channel (well i tried, with ending up in lots of trouble), because its depreciated in asterisk 1.4 and gone in 1.6. so i decided to do as proposed in UPGRADE.txt and asterisk-src/doc/queues-with-callback-members.txt, with one change, i'm not using the Local channel, because it is not showing the right status of the devices in the queue. (I wonder how the callcenter at digiums ist working with that). maybe anyone else having problems with queues in asterisk 1.4? yours christian gansberger ___ -- 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] Using a mobile phone via USB as an extension
Have you tried Fring? It's a softphone software for mobile phones http://www.fring.com/ Ish Nick Hill wrote: Thank you for the info Does anyone know if the cdc-modem interface which is available on mobile phones can actually potentially be used to initiate, or register for receiving a voice call? If so, I suppose USB 3G dongles could even be used as a voip-air interface! Would be interesting to find specs for these. Administrator TOOTAI wrote: Carlos Ruiz Diaz a écrit : Check chan_mobile. [...] Or use GSM gateway On Thu, Jul 2, 2009 at 3:20 PM, Nick Hill t...@nickhill.co.uk wrote: I have had a search for this, but didn't come up with any results, so maybe I am using the wrong terms, sorry if this is an FAQ. For those who want to forward their incoming voice calls to a mobile, it could be a cheaper option to call a mobile from another mobile on the same network. This probably wouldn't be useful for users in USA, Canada or Hong Kong as costs to call a mobile is the same as a land line. In other countries, it is very different. I know of a mobile operator who bundle lots of free on-network minutes with SIM cards. I wonder if it is possible to forward the call via a mobile phone tethered to an asterisk server through USB? Has anyone tried tethering a mobile phone to an asterisk server and configuring it as an asterisk extension so they can use free or cheap on-network minutes for the mobile leg of the call? ___ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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] Using a mobile phone via USB as an extension
Hello Ishfaq I have used Fring, but I don't believe it is capable of initiating GSM calls from VOIP. As I understand it, Fring does VOIP---Data bearer ---Fring-Microphone/speaker (wifi, 3G data) I am proposing \|/ | 3G/GSM---Asterisk the bit can be achieved now using Bluetooth with specific bluetooth dongles and specific handsets. USB especially using a USB 3G/GSM dongle would be a neater, and potentially more reliable solution. If there were standard protocols which can be used over the USB GSM modem interface which can be used to set the device to receive voice calls and send them over the USB interface, and to establish a voice call via the USB interface, this could potentially be a neat solution. However, I guess that even if a lot of equipment nominally supports such an interface, it may not be well tested. Interesting for a hack. Ishfaq Malik wrote: Have you tried Fring? It's a softphone software for mobile phones http://www.fring.com/ Ish ___ -- 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 possible to sen sms with asterisk in Spain?
Hi all, I´m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with the operators...) I have Elastix 1.3.2 and I have seen this url: http://mirror.su.lt/voip-info/wiki/view/Asterisk+cmd+Sms.html I have tried the smsq command but I can get it work, (as I say I´m a begginer and I don´t understand several concepts in this web) Can anyone help me with this? Thanks in advance ESG ___ -- 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] Rtp keepalive
Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all combinations of nat and qualify for the peer that has problems - rtp comfort noise is simply not sent. After trying to make it work for a day or so, I reported it as a bug (https://issues.asterisk.org/view.php?id=15466) but maybe someone here has some ideas how to make it work? -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 851 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com Gradwell - Internet for Business People Phone Services | Business Broadband | Email Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ -- 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] CIDlookup
On Thu, Jul 9, 2009 at 3:01 AM, Sriramd_r_sri...@hotmail.com wrote: Hi List I've a CID lookup hooked onto an inbound route (i m using trixbox) it runs well but it returns the value as CIDNAMECIDNUMBER ... if i just want to display the CIDNAME [leaving the quotes and CIDNUMBER] .. how can i do it ? do i have to edit some macro in extensions.conf ? rgds Sriram ___ Use Cut() http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cut -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] is possible to sen sms with asterisk in Spain?
Hi use www.kannel.org On Thu, Jul 9, 2009 at 3:26 PM, ESGLinux esggru...@gmail.com wrote: Hi all, I´m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with the operators...) I have Elastix 1.3.2 and I have seen this url: http://mirror.su.lt/voip-info/wiki/view/Asterisk+cmd+Sms.html I have tried the smsq command but I can get it work, (as I say I´m a begginer and I don´t understand several concepts in this web) Can anyone help me with this? Thanks in advance ESG ___ -- 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] is possible to sen sms with asterisk in Spain?
Same question here : How about in Belgium ?? Because core show application like sms gives information about the UK. Jonas. On Thu, 2009-07-09 at 11:26 +0200, ESGLinux wrote: Hi all, I´m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with the operators...) I have Elastix 1.3.2 and I have seen this url: http://mirror.su.lt/voip-info/wiki/view/Asterisk+cmd+Sms.html I have tried the smsq command but I can get it work, (as I say I´m a begginer and I don´t understand several concepts in this web) Can anyone help me with this? Thanks in advance ESG ___ -- 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] Restarting of B-channel on span 1
Hi Sir, I just want to confirm that it is located in zaptel.com or zapata.conf? because i have find resetinterval in zapta.conf but not in zaptel.conf.. Thanks _ cricket and news. Logon to MSN Video for the latest clips http://www.exploremyway.com___ -- 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] is possible to sen sms with asterisk in Spain?
Hi thanks for your answer, I don´t want to install code in the machine with the asterisk, (I have tried and I have dependencies that I can´t solve) so, the real question is with the software that comes with my elastix release can I send sms? thanks again, ESG 2009/7/9 Shahid Tel shahed...@gmail.com Hi use www.kannel.org On Thu, Jul 9, 2009 at 3:26 PM, ESGLinux esggru...@gmail.com wrote: Hi all, I´m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with the operators...) I have Elastix 1.3.2 and I have seen this url: http://mirror.su.lt/voip-info/wiki/view/Asterisk+cmd+Sms.html I have tried the smsq command but I can get it work, (as I say I´m a begginer and I don´t understand several concepts in this web) Can anyone help me with this? Thanks in advance ESG ___ -- 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] Failed to read gains: Invalid argument
Hi, all , hope u all are good and fine . me getting new error which i am pasting below.. This will came when i am reloading the asterisk. i also tried [reload chan_zap.so ] on asterisk cli, then out is same as i mention below, i there is any misconfiguration in zapata.conf?? i am posting below mine zapta.conf. hope that we help u to solve it . Thanks a lot... aman DhallY [Jul 9 14:52:41] WARNING[32279] chan_zap.c: Ignoring signalling [Jul 9 14:52:41] WARNING[32279] chan_zap.c: Ignoring rxwink [Jul 9 14:52:41] WARNING[32279] chan_zap.c: Ignoring switchtype [Jul 9 14:52:41] WARNING[32279] chan_zap.c: Ignoring signalling [Jul 9 14:52:41] WARNING[32279] chan_zap.c: Ignoring pridialplan [Jul 9 14:52:41] WARNING[32279] chan_zap.c: Ignoring resetinterval [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 1, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 2, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 3, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 4, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 5, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 6, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 7, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 8, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 9, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 10, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 11, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 12, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 13, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 14, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 15, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 17, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 18, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 19, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 20, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 21, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 22, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 23, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 24, ISDN PRI signalling [Jul 9 14:52:41] DEBUG[32279] chan_zap.c: Failed to read gains: Invalid argument [Jul 9 14:52:41] VERBOSE[32279] logger.c: -- Reconfigured channel 25, ISDN PRI signalling [Jul 9 14:52:41]
Re: [asterisk-users] Restarting of B-channel on span 1
resetinterval=never in zapata.conf.
you may want to reset them though, just not as frequently. The
resetinterval can take an integer as well.
Thanks,
Steve Totaro
On Wed, Jul 8, 2009 at 9:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI
Channels are restating after every interval of one hour, and i have search
and psot on
fourms and everyone said that this is a normal behaviour.
If this is a normal behaviour is there is any way to stop it { i still don't
know what is the reson to restart ever hour } . Because this is listed
everywhere that this is a normal behaviour, but not one mention {may be i am
not able to find it is listed some where} why this is nesessary? and if this
is not nessary how to stop it...
I think we all already know the message , but posting it for future
reference..
Thanks a lot .
Aman Dhally
--
ul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Event Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Queue Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: -- Remote UNIX connection
disconnected
[Jul 8 04:51:30] VERBOSE[3300] logger.c: -- B-channel 0/1 successfully
restarted on span 1
[Jul 8 04:51:35] VERBOSE[3300] logger.c: -- B-channel 0/2 successfully
restarted on span 1
[Jul 8 04:51:40] VERBOSE[3300] logger.c: -- B-channel 0/3 successfully
restarted on span 1
[Jul 8 04:51:45] VERBOSE[3300] logger.c: -- B-channel 0/4 successfully
restarted on span 1
[Jul 8 04:51:50] VERBOSE[3300] logger.c: -- B-channel 0/5 successfully
restarted on span 1
[Jul 8 04:51:55] VERBOSE[3300] logger.c: -- B-channel 0/6 successfully
restarted on span 1
[Jul 8 04:52:00] VERBOSE[3300] logger.c: -- B-channel 0/7 successfully
restarted on span 1
[Jul 8 04:52:05] VERBOSE[3300] logger.c: -- B-channel 0/8 successfully
restarted on span 1
[Jul 8 04:52:10] VERBOSE[3300] logger.c: -- B-channel 0/9 successfully
restarted on span 1
[Jul 8 04:52:15] VERBOSE[3300] logger.c: -- B-channel 0/10 successfully
restarted on span 1
[Jul 8 04:52:20] VERBOSE[3300] logger.c: -- B-channel 0/11 successfully
restarted on span 1
[Jul 8 04:52:25] VERBOSE[3300] logger.c: -- B-channel 0/12 successfully
restarted on span 1
[Jul 8 04:52:30] VERBOSE[3300] logger.c: -- B-channel 0/13 successfully
restarted on span 1
[Jul 8 04:52:35] VERBOSE[3300] logger.c: -- B-channel 0/14 successfully
restarted on span 1
[Jul 8 04:52:40] VERBOSE[3300] logger.c: -- B-channel 0/15 successfully
restarted on span 1
[Jul 8 04:52:45] VERBOSE[3300] logger.c: -- B-channel 0/17 successfully
restarted on span 1
[Jul 8 04:52:50] VERBOSE[3300] logger.c: -- B-channel 0/18 successfully
restarted on span 1
[Jul 8 04:52:55] VERBOSE[3300] logger.c: -- B-channel 0/19 successfully
restarted on span 1
[Jul 8 04:53:00] VERBOSE[3300] logger.c: -- B-channel 0/20 successfully
restarted on span 1
[Jul 8 04:53:05] VERBOSE[3300] logger.c: -- B-channel 0/21 successfully
restarted on span 1
[Jul 8 04:53:10] VERBOSE[3300] logger.c: -- B-channel 0/22 successfully
restarted on span 1
[Jul 8 04:53:15] VERBOSE[3300] logger.c: -- B-channel 0/23 successfully
restarted on span 1
[Jul 8 04:53:20] VERBOSE[3300] logger.c: -- B-channel 0/24 successfully
restarted on span 1
[Jul 8 04:53:25] VERBOSE[3300] logger.c: -- B-channel 0/25 successfully
restarted on span 1
[Jul 8 04:53:30] VERBOSE[3300] logger.c: -- B-channel 0/26 successfully
restarted on span 1
[Jul 8 04:53:35] VERBOSE[3300] logger.c: -- B-channel 0/27 successfully
restarted on span 1
[Jul 8 04:53:40] VERBOSE[3300] logger.c: -- B-channel 0/28 successfully
restarted on span 1
[Jul 8 04:53:45] VERBOSE[3300] logger.c: -- B-channel 0/29 successfully
restarted on span 1
[Jul 8 04:53:50] VERBOSE[3300] logger.c: -- B-channel 0/30 successfully
restarted on span 1
[Jul 8 04:53:55] VERBOSE[3300] logger.c: -- B-channel 0/31 successfully
restarted on span 1
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Re: [asterisk-users] is possible to sen sms with asterisk in Spain?
Am Donnerstag, den 09.07.2009, 11:26 +0200 schrieb ESGLinux: Hi all, I´m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with the operators...) Hi, the SMS code in Asterisk is - afaik - only for the landline type of SMS. It can behave as landline-SMS capable phone (like some of the Siemens Gigaset DECT devices, for example) and talk to a landline-SMS center that will for a certain charge forward short messages to mobile phones. It can also behave as landline-SMS center and talk to appropriate phones. As a background info, landline phones can recognize that a landline SMS center is calling them by caller ID (which must be programmed, many phones ship with the local companies' numbers preprogrammed) and will not ring the bell but silently answer the line. The message transfer works with 1200 baud modem-like analogue audio (even if the phone is an ISDN device) - you can watch the actual message bytes on the Asterisk CLI if you turn on debug, in some kind of simple protocol and some 8bit-to-7bit mapping. It cannot directly talk to mobile phones: short messages are transmitted out-of-band in the GSM networks, and the mobile operators will not allow you direct access there. After all, short messages make a hefty percentage of their income at a minimum percentage of infrastructure usage. The situation in Germany (and to my knowledge, in several other European states) is that you can connect to a premium-rate landline-SMS center and hand them a short message for relaying. As that is bound to cost hardly less than using a mobile phone directly, it is not at all interesting for me (ymmv). I prefer using one of those web-interface-to-sms providers (mine can be used with wget from scripts etc) and pay between 3 and 12 cents per message, depending on destination country and quality of service selection. They have been reliable for quite some time now, and I remember that landline-SMS was a little too fiddly for my taste. Regards Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restarting of B-channel on span 1
Bah, my mistake, as Steve said the entry goes in zapata.conf.
On 09/07/2009, Steve Totaro stot...@asteriskhelpdesk.com wrote:
resetinterval=never in zapata.conf.
you may want to reset them though, just not as frequently. The
resetinterval can take an integer as well.
Thanks,
Steve Totaro
On Wed, Jul 8, 2009 at 9:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI
Channels are restating after every interval of one hour, and i have search
and psot on
fourms and everyone said that this is a normal behaviour.
If this is a normal behaviour is there is any way to stop it { i still
don't
know what is the reson to restart ever hour } . Because this is listed
everywhere that this is a normal behaviour, but not one mention {may be i
am
not able to find it is listed some where} why this is nesessary? and if
this
is not nessary how to stop it...
I think we all already know the message , but posting it for future
reference..
Thanks a lot .
Aman Dhally
--
ul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Event Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Queue Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: -- Remote UNIX connection
disconnected
[Jul 8 04:51:30] VERBOSE[3300] logger.c: -- B-channel 0/1 successfully
restarted on span 1
[Jul 8 04:51:35] VERBOSE[3300] logger.c: -- B-channel 0/2 successfully
restarted on span 1
[Jul 8 04:51:40] VERBOSE[3300] logger.c: -- B-channel 0/3 successfully
restarted on span 1
[Jul 8 04:51:45] VERBOSE[3300] logger.c: -- B-channel 0/4 successfully
restarted on span 1
[Jul 8 04:51:50] VERBOSE[3300] logger.c: -- B-channel 0/5 successfully
restarted on span 1
[Jul 8 04:51:55] VERBOSE[3300] logger.c: -- B-channel 0/6 successfully
restarted on span 1
[Jul 8 04:52:00] VERBOSE[3300] logger.c: -- B-channel 0/7 successfully
restarted on span 1
[Jul 8 04:52:05] VERBOSE[3300] logger.c: -- B-channel 0/8 successfully
restarted on span 1
[Jul 8 04:52:10] VERBOSE[3300] logger.c: -- B-channel 0/9 successfully
restarted on span 1
[Jul 8 04:52:15] VERBOSE[3300] logger.c: -- B-channel 0/10 successfully
restarted on span 1
[Jul 8 04:52:20] VERBOSE[3300] logger.c: -- B-channel 0/11 successfully
restarted on span 1
[Jul 8 04:52:25] VERBOSE[3300] logger.c: -- B-channel 0/12 successfully
restarted on span 1
[Jul 8 04:52:30] VERBOSE[3300] logger.c: -- B-channel 0/13 successfully
restarted on span 1
[Jul 8 04:52:35] VERBOSE[3300] logger.c: -- B-channel 0/14 successfully
restarted on span 1
[Jul 8 04:52:40] VERBOSE[3300] logger.c: -- B-channel 0/15 successfully
restarted on span 1
[Jul 8 04:52:45] VERBOSE[3300] logger.c: -- B-channel 0/17 successfully
restarted on span 1
[Jul 8 04:52:50] VERBOSE[3300] logger.c: -- B-channel 0/18 successfully
restarted on span 1
[Jul 8 04:52:55] VERBOSE[3300] logger.c: -- B-channel 0/19 successfully
restarted on span 1
[Jul 8 04:53:00] VERBOSE[3300] logger.c: -- B-channel 0/20 successfully
restarted on span 1
[Jul 8 04:53:05] VERBOSE[3300] logger.c: -- B-channel 0/21 successfully
restarted on span 1
[Jul 8 04:53:10] VERBOSE[3300] logger.c: -- B-channel 0/22 successfully
restarted on span 1
[Jul 8 04:53:15] VERBOSE[3300] logger.c: -- B-channel 0/23 successfully
restarted on span 1
[Jul 8 04:53:20] VERBOSE[3300] logger.c: -- B-channel 0/24 successfully
restarted on span 1
[Jul 8 04:53:25] VERBOSE[3300] logger.c: -- B-channel 0/25 successfully
restarted on span 1
[Jul 8 04:53:30] VERBOSE[3300] logger.c: -- B-channel 0/26 successfully
restarted on span 1
[Jul 8 04:53:35] VERBOSE[3300] logger.c: -- B-channel 0/27 successfully
restarted on span 1
[Jul 8 04:53:40] VERBOSE[3300] logger.c: -- B-channel 0/28 successfully
restarted on span 1
[Jul 8 04:53:45] VERBOSE[3300] logger.c: -- B-channel 0/29 successfully
restarted on span 1
[Jul 8 04:53:50] VERBOSE[3300] logger.c: -- B-channel 0/30 successfully
restarted on span 1
[Jul 8 04:53:55] VERBOSE[3300] logger.c: -- B-channel 0/31 successfully
restarted on span 1
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Re: [asterisk-users] Using a mobile phone via USB as an extension
2009/7/2 Administrator TOOTAI ad...@tootai.net Carlos Ruiz Diaz a écrit : Check chan_mobile. [...] Or use GSM gateway Using a GSM gateway is possible but it's quite different as you need to insert a SIM card inside to let it work. ___ -- 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] Using a mobile phone via USB as an extension
2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com Check chan_mobile. Now is mature enough to be used in a server with low CPS. The USB connectivity will be introduced in the close future (I think) but by now it can be connected via bluetooth device. Where did you get this info (USB connectivity for chan_mobile) ? Is there a way to learn a bit more ? ___ -- 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] is possible to sen sms with asterisk in Spain?
2009/7/9 Anselm Martin Hoffmeister ans...@hoffmeister-online.de Am Donnerstag, den 09.07.2009, 11:26 +0200 schrieb ESGLinux: Hi all, I´m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with the operators...) Hi, the SMS code in Asterisk is - afaik - only for the landline type of SMS. It can behave as landline-SMS capable phone (like some of the Siemens Gigaset DECT devices, for example) and talk to a landline-SMS center that will for a certain charge forward short messages to mobile phones. It can also behave as landline-SMS center and talk to appropriate phones. As a background info, landline phones can recognize that a landline SMS center is calling them by caller ID (which must be programmed, many phones ship with the local companies' numbers preprogrammed) and will not ring the bell but silently answer the line. The message transfer works with 1200 baud modem-like analogue audio (even if the phone is an ISDN device) - you can watch the actual message bytes on the Asterisk CLI if you turn on debug, in some kind of simple protocol and some 8bit-to-7bit mapping. It cannot directly talk to mobile phones: short messages are transmitted out-of-band in the GSM networks, and the mobile operators will not allow you direct access there. After all, short messages make a hefty percentage of their income at a minimum percentage of infrastructure usage. The situation in Germany (and to my knowledge, in several other European states) is that you can connect to a premium-rate landline-SMS center and hand them a short message for relaying. As that is bound to cost hardly less than using a mobile phone directly, it is not at all interesting for me (ymmv). I prefer using one of those web-interface-to-sms providers (mine can be used with wget from scripts etc) and pay between 3 and 12 cents per message, depending on destination country and quality of service selection. They have been reliable for quite some time now, and I remember that landline-SMS was a little too fiddly for my taste. Regards Anselm ok thanks for your answer, I think your are right with the landline-SMS, Now my question changes to, how can I send a SMS to my cellular phone, what hardware, software, subcription to service or somthing else do I need? Thanks in advance ESG ___ -- 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] Using a mobile phone via USB as an extension
I read it in this list. I buit an application on top of chan_mobile and i needed usb connectivity to improve the bandwidth so i googled for the answer and one of the hits was from here. On 7/9/09, Olivier oza-4...@myamail.com wrote: 2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com Check chan_mobile. Now is mature enough to be used in a server with low CPS. The USB connectivity will be introduced in the close future (I think) but by now it can be connected via bluetooth device. Where did you get this info (USB connectivity for chan_mobile) ? Is there a way to learn a bit more ? -- Sent from Gmail for mobile | mobile.google.com ___ -- 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] Setting up a secure AMI?
Hi All, I've just upgraded our CRM and it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned regarding having an open channel between the two machines and there is scant information about setting up a more secure connection. Can anyone offer any good links or howtos for this? The CRM is vtiger and I couldn't see any references to ssl in the php code. TIA Alan ___ -- 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] Setting up a secure AMI?
On 10/7/09 12:05 AM, Alan Lord (News) wrote: Hi All, I've just upgraded our CRM and it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned regarding having an open channel between the two machines and there is scant information about setting up a more secure connection. Can anyone offer any good links or howtos for this? The CRM is vtiger and I couldn't see any references to ssl in the php code. You might want to have a look at OpenVPN -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] q: which Browser-GUI do u guys use?
While I agree with Steve on a philosophical level, there are a lot of merits to command lines and direct editing of configuration, there also comes a time when just getting the job done is benefited by a nice point-n-click. I have found in my career that I may spend a month neck deep in a project, such as implementing Asterisk, then for the following 6 months never have to touch it again. During those 6 months away, I would have been implementing a new intrusion prevention system, probably doing a bit of programming, managing my 300+ Linux servers, or helping our DBA setup new MS-SQL clusters. When I'm asked to do something like, say reroute all incoming calls through a new IVR with several new queues, it sure helps to have a gui to help out instead of having to relearn the guts of the system. But these are just my thoughts on the subject. And so far during my month of being neck deep in implementing Asterisk I have used FreePBX. Jeremy None. I'm a command line weenie. ) GUIs don't let you annotate your changes -- who did what (or what they thought they were doing), when, and why. ) GUIs don't support any sort of versioning. ) GUIs don't support any sort of configuration rollback. All of these are essential when something that used to work suddenly doesn't. (Sometimes, client's don't notice something isn't working for months -- way beyond my short term memory.) I'm sure I could come up with dozens more, these were just the first 3. (Probably not even the most important 3.) Oh. Here's 1 more -- GUIs impede truly understanding a system. ___ -- 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] q: port forwarding or NAT
hi, making may way through all this...internal sip registration works,(cant call yet but anyhow)... the asterisk box is obvisoulsy behind a router. im not 100% sure if i should go with port forwarding or NAT and if a or b, what additional setup is actually correct? sip_nat.conf # this is when i got the NAT -route, right? #gets all the dyndns-stuff #externip = home.mydomain.com (Enter your DynamicDNS domain name. Obviously it's just easier to get a static IP address and avoid using DynamicDNS altogether.) externhost = home.mydomain.com externrefresh = 5 (which means lookup hostname every 5 minutes to refresh ip adress) localnet = internal.network.address.0/255.255.255.0 thx ___ -- 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] Setting up a secure AMI?
On 9 Jul 2009, at 13:05, Alan Lord (News) wrote: Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned regarding having an open channel between the two machines and there is scant information about setting up a more secure connection. Can anyone offer any good links or howtos for this? You can probably tunnel it over SSH. Steve ___ -- 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] Using a mobile phone via USB as an extension
Just google/bing it. http://voip-info.org/wiki/view/chan_mobile On Thu, Jul 9, 2009 at 12:56 PM, Olivier oza-4...@myamail.com wrote: 2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com Check chan_mobile. Now is mature enough to be used in a server with low CPS. The USB connectivity will be introduced in the close future (I think) but by now it can be connected via bluetooth device. Where did you get this info (USB connectivity for chan_mobile) ? Is there a way to learn a bit more ? ___ -- 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] Setting up a secure AMI?
On 09/07/09 14:40, Steve Howes wrote: On 9 Jul 2009, at 13:05, Alan Lord (News) wrote: Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned regarding having an open channel between the two machines and there is scant information about setting up a more secure connection. Can anyone offer any good links or howtos for this? You can probably tunnel it over SSH. Yes, I am trying to set up a [simple] stunnel connection. Nearly there... If anyone has a decent how to I'd live to have a link :-) Alan ___ -- 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] Queue autopause
Christian Gansberger escribió:
On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co wrote:
Christian Gansberger escribió:
Hi all!
I want to autopause my queue member when they are not answering within
20 seconds, and the autopause
should affect all queues they are member of, not only the queue where
the call was not answered.
Is there a way to do that?
The members gets dynamically added. I'm using asterisk 1.4.21.2.
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Why would you want to do that? The purpose of the autopause is to
discard the absent agent that is not responding to the calls to not
try it anymore until it gets unpaused by a supervisor or someone else,
and therefore the pause is made to all queues the agent is member of.
Why pause it on only one queue, letting it ring on other queues?
Aside from the purpose you have on this, I think you would need to
modify the app_queue.c code to make the parameter configurable inside
each queue definition and not on the general section of queues.conf.
Then you would need to modify the logic to handle the autopause
configured for each queue. This is a general idea as I didn't take a
deep look of app_queue.c to see how it works exactly.
Any other solution without changing asterisk code would imply a external
application that monitors the queues and makes the custom autopause you
need.
Just my two cents...
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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To make things clearer:
I want the queue member is autopaused on all queues. As a matter of
fact in asterisk (vers. 1.4.24.1)
the queue member is only paused on one Queue.
I tried setting autopause=yes in general context, which doesn't do anything.
So i set autopause=yes in every Queue definition, which is working,
but only on that queue.
I don't use the agents channel (well i tried, with ending up in lots
of trouble), because its
depreciated in asterisk 1.4 and gone in 1.6. so i decided to
do as proposed in UPGRADE.txt and
asterisk-src/doc/queues-with-callback-members.txt,
with one change, i'm not using the Local channel, because it is not
showing the right status
of the devices in the queue. (I wonder how the callcenter at digiums
ist working with that).
maybe anyone else having problems with queues in asterisk 1.4?
yours
christian gansberger
You're right, the autopause on its standard behavior pauses only the
member of the queue where it belongs. Taking a little look at
app_queue.c (http://www.asterisk.org/doxygen/1.4/app__queue_8c.html) you
can very easily patch the source code to achieve the functionality you
want. The key functions are:
static int set_member_paused - Traverses the queues doing all the
things necessary on all different scenarios (realtime, etc) to pause the
member you give to it. If there's no queue name given, it with pause the
member on all queues (the PAUSEALL event).
static void rna - (as the doxygen doc says) RNA == Ring No Answer.
Common code that is executed when we try a queue member and they don't
answer.
If you take a look to the rna function, with autopaused enabled it will
pause the member if it doesn't answer the queue call after the timeout
time. You can make it pause all members just by changing this one line:
02164 if (!set_member_paused
http://www.asterisk.org/doxygen/1.4/app__queue_8c.html#d61f43e341bcf4c523f2fdb01ece066b(qe-parent
http://www.asterisk.org/doxygen/1.4/structqueue__ent.html#59ceee334ec79ed344313a7e8affb3fc-name
http://www.asterisk.org/doxygen/1.4/structcall__queue.html#188159d17b341b26fcfe4b57baefd372,
interface
http://www.asterisk.org/doxygen/1.4/structcallattempt.html#8ee1350d5c943c7ee1ad3da9078eda25, 1)) {
to
02164 if (!set_member_paused
http://www.asterisk.org/doxygen/1.4/app__queue_8c.html#d61f43e341bcf4c523f2fdb01ece066b(,
interface
http://www.asterisk.org/doxygen/1.4/structcallattempt.html#8ee1350d5c943c7ee1ad3da9078eda25,
1)) {
That way we don't send the queue name, pausing it in all the queues it
is member of.
Although it's not tested, it might work for you. That's the beauty of
Asterisk and well documented Open Source projects, you can get to the
code as deep as you want, learn from it how it works, and change
it/improve it according to your needs. Good contributions make it to the
official code as well.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] CIDlookup
Use CALLERID(name). http://www.voip-info.org/wiki/view/Asterisk+func+callerid Steve Totaro wrote: On Thu, Jul 9, 2009 at 3:01 AM, Sriramd_r_sri...@hotmail.com wrote: Hi List I've a CID lookup hooked onto an inbound route (i m using trixbox) it runs well but it returns the value as CIDNAMECIDNUMBER ... if i just want to display the CIDNAME [leaving the quotes and CIDNUMBER] .. how can i do it ? do i have to edit some macro in extensions.conf ? rgds Sriram ___ Use Cut() http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cut ___ -- 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] is possible to sen sms with asterisk in Spain?
ESGLinux wrote: 2009/7/9 Anselm Martin Hoffmeister ans...@hoffmeister-online.de mailto:ans...@hoffmeister-online.de Am Donnerstag, den 09.07.2009, 11:26 +0200 schrieb ESGLinux: Hi all, I´m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with the operators...) Hi, the SMS code in Asterisk is - afaik - only for the landline type of SMS. It can behave as landline-SMS capable phone (like some of the Siemens Gigaset DECT devices, for example) and talk to a landline-SMS center that will for a certain charge forward short messages to mobile phones. It can also behave as landline-SMS center and talk to appropriate phones. As a background info, landline phones can recognize that a landline SMS center is calling them by caller ID (which must be programmed, many phones ship with the local companies' numbers preprogrammed) and will not ring the bell but silently answer the line. The message transfer works with 1200 baud modem-like analogue audio (even if the phone is an ISDN device) - you can watch the actual message bytes on the Asterisk CLI if you turn on debug, in some kind of simple protocol and some 8bit-to-7bit mapping. It cannot directly talk to mobile phones: short messages are transmitted out-of-band in the GSM networks, and the mobile operators will not allow you direct access there. After all, short messages make a hefty percentage of their income at a minimum percentage of infrastructure usage. The situation in Germany (and to my knowledge, in several other European states) is that you can connect to a premium-rate landline-SMS center and hand them a short message for relaying. As that is bound to cost hardly less than using a mobile phone directly, it is not at all interesting for me (ymmv). I prefer using one of those web-interface-to-sms providers (mine can be used with wget from scripts etc) and pay between 3 and 12 cents per message, depending on destination country and quality of service selection. They have been reliable for quite some time now, and I remember that landline-SMS was a little too fiddly for my taste. Regards Anselm ok thanks for your answer, I think your are right with the landline-SMS, Now my question changes to, how can I send a SMS to my cellular phone, what hardware, software, subcription to service or somthing else do I need? Thanks in advance ESG Take a look at: http://www.ozekisms.com/index.php?owpn=319 See Kannel as well: http://www.kannel.org/ Jorge Mendoza ___ -- 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] Weird audio problem with remote IVRs + DMTF
Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in dead air. With the one-way audio issue, is it possible that something has locked the channel into some mode where all audio being sent is muted? (As a result of DTMF?) I'm really perplexed by this one. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using a mobile phone via USB as an extension
Hello Sasa The page you point to doesn't talk about USB connectivity for chan_mobile. It does talk about bluetooth connectivity, which can be achieved by way of a USB bluetooth dongle, but that is not the same thing. I am talking about using standard interfaces exposed by mobile devices (mobile phones or USB dongles), exposed via USB, to establish voice calls as an extension to asterisk. And examining whether the standard CDC modem interface or any other standard interface generally carries the required commands. There are several potential benefits to using the USB approach: 1) Devices will be immune to interference from other 2.4Ghz devices. Bluetooth is prone. 2) Potentially better call quality. Using bluetooth, the call will need to be converted from the VOIP codec, to PCM then to the bluetooth codec then to the over-the-air mobile codec (GSM), with associated latency. There may be a possibility of cutting one of these conversions. 3) This could be used with a USB data dongle interface, which are cheap, relatively simple, and are powered from the USB port. They also generally offer great sensitivity over the air. 4) The bluetooth approach is dependent on a working (bug-free) bluetooth stack, with specific hardware, which may or may not be available. 5) less software and hardware needed in the chain, so potentially more maintainable and reliable. The bluetooth approach has some advantages: 1) The bluetooth interface is standardised and any device should theoretically work with any other of the given generation. The required codecs, for example, are well defined. (in practice, this isn't the case - for example, the Nokia E65 is bluetooth capable and doesn't work, and the chan_mobile is fussy about which bluetooth dongles you use). 2) Even if the USB interface does provide necessary interfaces for voice telephony, they may not be well tested. (buggy). We don't know - but then, the bluetooth approach is, in practice, hardware dependent also. 3) Bluetooth dongles are swallowed by a mobile phone. One dongle needed for each phone/channel. (they are cheap, but may have other ramifications). Someone may have investigated the USB approach already and discounted it. Or if not, it may be worth further examination. Carlos indicates that USB support may be available in chan_mobile but I can't find any references to it, and I think Oliver is looking for more info as well. Sasa Bobek wrote: Just google/bing it. http://voip-info.org/wiki/view/chan_mobile On Thu, Jul 9, 2009 at 12:56 PM, Olivier oza-4...@myamail.com mailto:oza-4...@myamail.com wrote: 2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com mailto:carlos.ruizd...@gmail.com Check chan_mobile. Now is mature enough to be used in a server with low CPS. The USB connectivity will be introduced in the close future (I think) but by now it can be connected via bluetooth device. Where did you get this info (USB connectivity for chan_mobile) ? Is there a way to learn a bit more ? ___ -- 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] Weird audio problem with remote IVRs + DMTF
On Thu, Jul 9, 2009 at 1:13 PM, James Lamannajlama...@gmail.com wrote: Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in dead air. With the one-way audio issue, is it possible that something has locked the channel into some mode where all audio being sent is muted? (As a result of DTMF?) I'm really perplexed by this one. Thanks. -- James Did you check something in features.conf? I suppose you don't have one way audio at other times? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Using a mobile phone via USB as an extension
Nick Hill escribio': Hello Sasa The page you point to doesn't talk about USB connectivity for chan_mobile. It does talk about bluetooth connectivity, which can be achieved by way of a USB bluetooth dongle, but that is not the same thing. I am talking about using standard interfaces exposed by mobile devices (mobile phones or USB dongles), exposed via USB, to establish voice calls as an extension to asterisk. And examining whether the standard CDC modem interface or any other standard interface generally carries the required commands. There are several potential benefits to using the USB approach: 1) Devices will be immune to interference from other 2.4Ghz devices. Bluetooth is prone. 2) Potentially better call quality. Using bluetooth, the call will need to be converted from the VOIP codec, to PCM then to the bluetooth codec then to the over-the-air mobile codec (GSM), with associated latency. There may be a possibility of cutting one of these conversions. 3) This could be used with a USB data dongle interface, which are cheap, relatively simple, and are powered from the USB port. They also generally offer great sensitivity over the air. 4) The bluetooth approach is dependent on a working (bug-free) bluetooth stack, with specific hardware, which may or may not be available. 5) less software and hardware needed in the chain, so potentially more maintainable and reliable. The bluetooth approach has some advantages: 1) The bluetooth interface is standardised and any device should theoretically work with any other of the given generation. The required codecs, for example, are well defined. (in practice, this isn't the case - for example, the Nokia E65 is bluetooth capable and doesn't work, and the chan_mobile is fussy about which bluetooth dongles you use). 2) Even if the USB interface does provide necessary interfaces for voice telephony, they may not be well tested. (buggy). We don't know - but then, the bluetooth approach is, in practice, hardware dependent also. 3) Bluetooth dongles are swallowed by a mobile phone. One dongle needed for each phone/channel. (they are cheap, but may have other ramifications). Someone may have investigated the USB approach already and discounted it. Or if not, it may be worth further examination. Carlos indicates that USB support may be available in chan_mobile but I can't find any references to it, and I think Oliver is looking for more info as well. Sasa Bobek wrote: Just google/bing it. http://voip-info.org/wiki/view/chan_mobile On Thu, Jul 9, 2009 at 12:56 PM, Olivier oza-4...@myamail.com mailto:oza-4...@myamail.com wrote: 2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com mailto:carlos.ruizd...@gmail.com Check chan_mobile. Now is mature enough to be used in a server with low CPS. The USB connectivity will be introduced in the close future (I think) but by now it can be connected via bluetooth device. Where did you get this info (USB connectivity for chan_mobile) ? Is there a way to learn a bit more ? ___ -- 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 There are some links about a module called chan_sebi that can use some Huawei USB dongles as a voice interface (in Spanish): http://odicha.wordpress.com/2009/06/30/chan_sebi-1-que-como-y-por-que/ http://odicha.wordpress.com/2009/06/30/chan_sebi-2-trasteando-con-el-codigo/ -- perl -e '$x=2.3;printf(%.0f + %.0f = %.0f\n,$x,$x,$x+$x);' ___ -- 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] setting up phones
Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. _ Lauren found her dream laptop. Find the PC that’s right for you. http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290___ -- 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] 1.6 macro deprecation, dial macros
I understand that standalone macros have been deprecated in 1.6 for gosub routines. I've been working on converting them all but was wondering about dial macros - it doesn't look like there's a replacement yet to call a gosub routine from the dial command. Or am I looking at this wrong? hose ___ -- 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] Connecting two Asterisk together via SIP + DISA
Hi all, I need to test the following scenario: +---+ +---+ | asterisk 1| | asterisk 2| +---+ +---+ | | | | ___|__|___ | | | | | | +---+ +---+ | ATA 1 | | ATA 2 | +---+ +---+ / \ / \ /\ /\ 21 22 1011 That is, I have 2 asterisks connected via SIP, two ATAs with two lines, and the ATA1 is registered with asterisk1 and ATA2 is registered with asterisk2, and all incoming calls in asterisk2 from the asterisk1 (via SIP), are answered by a DISA. I can make calls between ATA1 and ATA2 without problems (the call will be routed to the asterisk1 to asterisk2, falls in DISA and I call one of the phones ATA2). I am now trying to make the call coming from,eg, extension 21, go to the asterisk1 - asterisk2, answered by the DISA and go back asterisk1, ringing the branch 22. Since I am newbie in this matter, I wonder with friends from the list if this is possible ... Or is there another way to do this Below is my conf files. Rgs Cesar === asterisk 1 ** sip.conf [21] type=friend context=phones ; Where to start in the dialplan when this phone calls secret=21 ;callerid=John Doe 1234 ; Full caller ID, to override the phones config ; on incoming calls to Asterisk host=dynamic; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;canreinvite=yes; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a friend ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1...@default ; mailbox 1234 in voicemail context default disallow=all ; need to disallow=all before we can use allow= allow=ulaw ; Note: In user sections the order of codecs ; listed with allow= does NOT matter! allow=alaw allow=g723.1 ; Asterisk only supports g723.1 pass-thru! allow=g729 ; Pass-thru only unless g729 license obtained ;callingpres=allowed_passed_screen; Set caller ID presentation ; See doc/callingpres.txt for more information [22] type=friend context=phones ; Where to start in the dialplan when this phone calls secret=22 ;callerid=John Doe 1234 ; Full caller ID, to override the phones config ; on incoming calls to Asterisk host=dynamic; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;canreinvite=yes; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a friend ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1...@default ; mailbox
Re: [asterisk-users] setting up phones
It should be pretty simple. Follow the instructions on this page http://www.voiptalk.org/products/aastra-setup.html put the username from sip.conf into the first 4 fields, the secret into the password field and your asterisk ip into the fields that say voiptalk.org users.conf [207] username=207 transfer=yes mailbox=207 call-limit=2 fullname=mickey mouse registersip=no host=dynamic callgroup=1 context=default cid_number=207 hasvoicemail=yes vmsecret=1234 email=u...@yourpbx.com threewaycalling=yes hasdirectory=yes callwaiting=yes hasmanager=yes managerread=system,call,log,verbose,command,agent,user,config managerwrite=system,call,log,verbose,command,agent,user,config hasagent=yes hassip=yes hasiax=no secret=x nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 macaddress=001170 autoprov=yes label=207 linenumber=1 disallow=all allow=ulaw,gsm sip.conf [207] type=peer context=phones host=dynamic fromuser=207 call-limit=3 secret=x canreinvite=yes directrtpsetup=no nat=yes qualify=yes register = 207:xx...@yourpbx.com/207 defaultip=1.2.3.4 mailbox=207 disallow=all allow=alaw _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setting up phones Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. _ Lauren found her dream laptop. Find the PC that http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290 's right for you. ___ -- 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] PRI failover to SIP trunk
Hello, I've found a little documentation on voip-info and on the asterisk- users list, although I was hoping for an example of a tried-and-true failover setup between PRI and SIP. We are an outgoing call center that uses asterisk 1.4 connected to 2 PRIs from the local telephone company in one group (g1) and a SIP trunk from bandwidth.com. The PRIs are the primary outgoing service, however we have been experiencing some issues where one or both of them can fail randomly. We are working with the telephone company to have this resolved. In the meantime, we want to have a good failover solution where if both PRIs fail, asterisk will dial out through the SIP trunk. I've found solutions as simple as two Dial commands one after the other, and others where the failover Dial is in a jump to CONGESTION. Unfortunately we don't have a testing environment, so the solution really has to work. Does anyone else on the list have a PRI to VoIP failover setup that's worked for them in a high volume environment? Thanks! Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1 Rochester, NY 14624 Office: 888-865-0065 x202 Mobile: 585-705-1400 ___ -- 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] setting up phones
I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jul 2009 14:03:50 -0500 Subject: Re: [asterisk-users] setting up phones It should be pretty simple. Follow the instructions on this page http://www.voiptalk.org/products/aastra-setup.html put the username from sip.conf into the first 4 fields, the secret into the password field and your asterisk ip into the fields that say voiptalk.org users.conf [207] username=207 transfer=yes mailbox=207 call-limit=2 fullname=mickey mouse registersip=no host=dynamic callgroup=1 context=default cid_number=207 hasvoicemail=yes vmsecret=1234 email=u...@yourpbx.com threewaycalling=yes hasdirectory=yes callwaiting=yes hasmanager=yes managerread=system,call,log,verbose,command,agent,user,config managerwrite=system,call,log,verbose,command,agent,user,config hasagent=yes hassip=yes hasiax=no secret=x nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 macaddress=001170 autoprov=yes label=207 linenumber=1 disallow=all allow=ulaw,gsm sip.conf [207] type=peer context=phones host=dynamic fromuser=207 call-limit=3 secret=x canreinvite=yes directrtpsetup=no nat=yes qualify=yes register = 207:xx...@yourpbx.com/207 defaultip=1.2.3.4 mailbox=207 disallow=all allow=alaw From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setting up phones Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. Lauren found her dream laptop. Find the PC that’s right for you. _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009___ -- 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] setting up phones
What do you get from sip show peers in CLI? Do you have your ip address in sip.conf? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 4:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf [501] username = 501 transfer = yes mailbox = 501 call-limit = 100 type = peer fullname = 501 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 501 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 501 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 00085d10927f autoprov = yes label = 501 linenumber = 1 LINEKEYS = 1 [500] username = 500 transfer = yes mailbox = 500 call-limit = 100 type = peer fullname = 500 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 500 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 500 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 00085d1095aa autoprov = yes label = 500 linenumber = 1 LINEKEYS = 1 _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 9 Jul 2009 14:03:50 -0500 Subject: Re: [asterisk-users] setting up phones It should be pretty simple. Follow the instructions on this page http://www.voiptalk.org/products/aastra-setup.html put the username from sip.conf into the first 4 fields, the secret into the password field and your asterisk ip into the fields that say voiptalk.org users.conf [207] username=207 transfer=yes mailbox=207 call-limit=2 fullname=mickey mouse registersip=no host=dynamic callgroup=1 context=default cid_number=207 hasvoicemail=yes vmsecret=1234 email=u...@yourpbx.com threewaycalling=yes hasdirectory=yes callwaiting=yes hasmanager=yes managerread=system,call,log,verbose,command,agent,user,config managerwrite=system,call,log,verbose,command,agent,user,config hasagent=yes hassip=yes hasiax=no secret=x nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 macaddress=001170 autoprov=yes label=207 linenumber=1 disallow=all allow=ulaw,gsm sip.conf [207] type=peer context=phones host=dynamic fromuser=207 call-limit=3 secret=x canreinvite=yes directrtpsetup=no nat=yes qualify=yes register = 207:xx...@yourpbx.com/207 defaultip=1.2.3.4 mailbox=207 disallow=all allow=alaw _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Thursday, July 09, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setting up phones Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. _ Lauren found her dream laptop. Find the http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290 PC that's right for you. _ Windows LiveT: Keep your life in sync. Check it out. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009 ___ -- 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] 1.6 macro deprecation, dial macros
On Thursday 09 July 2009 14:13:28 Hose wrote: I understand that standalone macros have been deprecated in 1.6 for gosub routines. I've been working on converting them all but was wondering about dial macros - it doesn't look like there's a replacement yet to call a gosub routine from the dial command. Or am I looking at this wrong? Look at the U option. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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] PRI failover to SIP trunk
On Thu, Jul 9, 2009 at 4:37 PM, Jason Martin jmar...@metrixmatrix.comwrote:
Hello,
I've found a little documentation on voip-info and on the asterisk-
users list, although I was hoping for an example of a tried-and-true
failover setup between PRI and SIP.
We are an outgoing call center that uses asterisk 1.4 connected to 2
PRIs from the local telephone company in one group (g1) and a SIP
trunk from bandwidth.com. The PRIs are the primary outgoing service,
however we have been experiencing some issues where one or both of
them can fail randomly. We are working with the telephone company to
have this resolved.
In the meantime, we want to have a good failover solution where if
both PRIs fail, asterisk will dial out through the SIP trunk. I've
found solutions as simple as two Dial commands one after the other,
and others where the failover Dial is in a jump to CONGESTION.
Unfortunately we don't have a testing environment, so the solution
really has to work.
Does anyone else on the list have a PRI to VoIP failover setup that's
worked for them in a high volume environment?
Thanks!
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office: 888-865-0065 x202
Mobile: 585-705-1400
Simple enough,
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
That is if Zap/DAHDI completely craps out. If the dialplan/Asterisk thinks
it is working it will hang.
If totally out of commission, then the second priority gets called.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] PRI failover to SIP trunk
On Thu, Jul 9, 2009 at 5:31 PM, Steve Totaro stot...@first-notification.com
wrote:
On Thu, Jul 9, 2009 at 4:37 PM, Jason Martin jmar...@metrixmatrix.comwrote:
Hello,
I've found a little documentation on voip-info and on the asterisk-
users list, although I was hoping for an example of a tried-and-true
failover setup between PRI and SIP.
We are an outgoing call center that uses asterisk 1.4 connected to 2
PRIs from the local telephone company in one group (g1) and a SIP
trunk from bandwidth.com. The PRIs are the primary outgoing service,
however we have been experiencing some issues where one or both of
them can fail randomly. We are working with the telephone company to
have this resolved.
In the meantime, we want to have a good failover solution where if
both PRIs fail, asterisk will dial out through the SIP trunk. I've
found solutions as simple as two Dial commands one after the other,
and others where the failover Dial is in a jump to CONGESTION.
Unfortunately we don't have a testing environment, so the solution
really has to work.
Does anyone else on the list have a PRI to VoIP failover setup that's
worked for them in a high volume environment?
Thanks!
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office: 888-865-0065 x202
Mobile: 585-705-1400
Simple enough,
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
That is if Zap/DAHDI completely craps out. If the dialplan/Asterisk thinks
it is working it will hang.
If totally out of commission, then the second priority gets called.
Let me clarify that I think that is how it works. Been a long time.
Maybe it was the old N+101 trick? Not sure why that was ever deprecated.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] setting up phones
On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote: I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf Either you did not explain your network topology very well or that is your problem. Unless you are trying to segregate your VoIP traffic, plug everything into the switch. If using DHCP, get the IP and try pinging the phones from the Asterisk box. I bet it is just a network issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 Segmentation Faults Using Skinny (v1.6.0.10)
Hi all,
I've just built a new installation of CentOS release 5.3 (Final) and
have installed both
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gzAsterisk
1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe
trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing
complex - Pentium Dual core 2ghz - 1gb ram - 70gb sata hd).
The setup at this point is real simple with one Cisco 7960 phone
registering with Asterisk using Skinny.
I'm finding that simple things as pressing any of the buttons on the
phone is enough to cause Asterisk to randomly restart from a
segmentation fault.
I've tried this with 1.6.1.1 and, after recompiling and replacing, 1.6.0.10.
I followed
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
as a basis for installation leaving out things I didnt want to set up
(odbc / web admin ).
The only thing that didn't seem to go too well was the setup Dahdi
(dahdi-linux-2.2.0.1). Although I can do a 'make' and 'make install',
'make config' didnt work and there are no etc/dahdi/ directory to change
any config files (as suggested by the guide). This may not be related
but just in case I thought I would mention it.
This is from the console after pressing the 'speaker' button a couple of
times.
/usr/sbin/safe_asterisk: line 146: 21513 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal EXITSTATUS-128.
Automatically restarting Asterisk.
If I don't use the phone, Asterisk will stay running.
I can dial the 1000 test extension along with the 500 inter-asterisk
test, these seem to work as expected as long as I dial the number and
hit 'dial' on the phone rather than selecting the line and trying to
dial each digit in turn. If I try that then at some random point (but
not always) Asterisk will fault.
The firmware version on the phone is 7.2 to which I've had this phone
and several others running off a 1.2 setup for years (using
chan_skinny?) but thought it time to update Asterisk.
Anyone have any pointers please on what to check next?
Thanks,
Wayne
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[asterisk-users] can 2 quad T1 cards work in 1 quad core amd server
I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Basically used to dial out and deliver messages. play wav files for the message. Any thoughts. Jerry ___ -- 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] 1.6 macro deprecation, dial macros
What you say...Tilghman Lesher (tilgh...@mail.jeffandtilghman.com): On Thursday 09 July 2009 14:13:28 Hose wrote: I understand that standalone macros have been deprecated in 1.6 for gosub routines. I've been working on converting them all but was wondering about dial macros - it doesn't look like there's a replacement yet to call a gosub routine from the dial command. Or am I looking at this wrong? Look at the U option. Thanks - it wasn't in any of the usual documentation places I source, but it was clearly in the application's internal docs (which I often forget to look at). hose ___ -- 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] can 2 quad T1 cards work in 1 quad core amd server
Jerry Geis wrote: oh you mean a telemarketing pest server ? I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Basically used to dial out and deliver messages. play wav files for the message. Any thoughts. Jerry ___ -- 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] can 2 quad T1 cards work in 1 quad core amd server
On Thu, 9 Jul 2009, Jerry Geis wrote: I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Basically used to dial out and deliver messages. play wav files for the message. Within the environment you described, yes. But why would you want to put 184 to 192 eggs in one basket? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com 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
Re: [asterisk-users] can 2 quad T1 cards work in 1 quad core amd server
On Thu, Jul 9, 2009 at 6:48 PM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 9 Jul 2009, Jerry Geis wrote: I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Basically used to dial out and deliver messages. play wav files for the message. Within the environment you described, yes. But why would you want to put 184 to 192 eggs in one basket? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 He could put a T1 failover box in front and have two identical servers :P I see your point, but I have 1.2.x boxen with eight T1 ports that have years of uptime. Just don't mess with them and make sure they have clean/constant power. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] can 2 quad T1 cards work in 1 quad core amd server
This is not a telemarkeing machine. Customer wants to be able to contact their own people with this (I dont ask why) I thought about a second machine and using SIP to connection back to the server. Is that a better solution? have 1 card in the server and another card in another machine running SIP trunk back? jerry ___ -- 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] can 2 quad T1 cards work in 1 quad core amd server
On Thu, Jul 9, 2009 at 6:55 PM, Jerry Geis ge...@pagestation.com wrote: This is not a telemarkeing machine. Customer wants to be able to contact their own people with this (I dont ask why) I thought about a second machine and using SIP to connection back to the server. Is that a better solution? have 1 card in the server and another card in another machine running SIP trunk back? jerry Redfone might make an 8 port. TDMoE would work but SIP works just fine. I have run a DS3 worth of traffic over SIP and I am sure many others have done much much more. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] can 2 quad T1 cards work in 1 quad core amd server
On Thu, Jul 9, 2009 at 6:34 PM, Jerry Geisge...@pagestation.com wrote: I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Yes. Buy a server that has the corresponding ports to accommodate the cards. A modern server is probably going to have PCI-E slots and you'll want the appropriate TDM cards. Any thoughts. Yes. That's a lot of power to drive a comparatively small number of calls. Also, I find it interesting that so many of the answers to these questions turn into a: 'you're going to use that for bad purposes' which is retorted with: 'no I'm not' I will say that I make a boatload of these outgoing 'play a file' calls, and they are for legitimate purposes to existing customers. Is it really that hard to imagine a business with a good reason to call their customers? I think that if somebody asked for a postal machine that processed a large number of letters somebody would say 'you're using that for junk mail', and somebody else would say 'fine, I won't mail you your paycheck'. I mostly just think it amusing that everybody has bad motives until proven otherwise. And moreover, the idea that you can somehow inoculate the world from people with bad motives if you don't provide assistance. ___ -- 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] can 2 quad T1 cards work in 1 quad core amd server
David Backeberg wrote: On Thu, Jul 9, 2009 at 6:34 PM, Jerry Geisge...@pagestation.com wrote: I was wondering if (2) quad T1 cards will work nicely in 1 server with a quad core AMD 3.0 gig cpu? Yes. Buy a server that has the corresponding ports to accommodate the cards. A modern server is probably going to have PCI-E slots and you'll want the appropriate TDM cards. Any thoughts. Yes. That's a lot of power to drive a comparatively small number of calls. Also, I find it interesting that so many of the answers to these questions turn into a: 'you're going to use that for bad purposes' which is retorted with: 'no I'm not' well just look at the fact that the cost to place a call is declining and the cost to mail a letter is increasing, then you see where the focus of abuse is going. I will say that I make a boatload of these outgoing 'play a file' calls, and they are for legitimate purposes to existing customers. Is it really that hard to imagine a business with a good reason to call their customers? I think that if somebody asked for a postal machine that processed a large number of letters somebody would say 'you're using that for junk mail', and somebody else would say 'fine, I won't mail you your paycheck'. I mostly just think it amusing that everybody has bad motives until proven otherwise. And moreover, the idea that you can somehow inoculate the world from people with bad motives if you don't provide assistance. ___ -- 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] Educational institutions: Your Asterisk experiences wanted!
Hello! I've been asked to get a show of hands for some analysts for users in Higher Ed - Universities, Colleges, or any other 2 or 4 year degree- granting institutions. If this fits you, please let me know your contact data and briefly how you're using Asterisk, and if you don't mind I can pass your contact data along (for consumption by humans only - this is not a mailing list or marketing list) to some analysts who might be interested in talking to you about your Open Source experience(s). Sound interesting? Mail me back and we can discuss in detail. Also, I've been asked by a community member to create an .edu mailing list. I'm all for this, and I've been swamped with other things to do and haven't gotten around to it yet. Would you be interested in such a list that is specifically for discussing implementation issues for such higher educational institutions? JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- 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
