[asterisk-users] Queue member status - BUSY
Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas? regards, ryan icasiano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing
You might have to tamper the main a2billing.php or more files for that feature to work. Or it might cost around $800 in development time. On Tue, Oct 19, 2010 at 4:34 AM, Baha @ SH i...@saudihome.com wrote: Exactly, I don’t want that, it’s annoying! I just want it to run if the customer balance reach for example 1 dollar! Anyway? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, October 18, 2010 8:17 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] a2billing I don't think voucher can be triggered to announce at certain threshold ONLY but it will be run everytime at the begining after PIN is asked for. By default it's set to: Press 8 to fill up with a voucher. System Settings is the last in the menu. -Bruce On Tue, Oct 19, 2010 at 2:31 AM, Baha @ SH i...@saudihome.com wrote: I am sorry , but where is System Settings??? And what is the parameter name? And also, id like to mention that the voucher is working, only when balance is below minimum balance it does not go to voucher ivr. Thanks, awaiting,,, *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, October 18, 2010 12:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] a2billing Turn on the voucher feature in System Settings and it will tell the user right after the PIN authentication or CLID authentication that their balance is below threshold and they should pay. -Bruce On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH i...@saudihome.com wrote: Not sure if a2billing can be shared here, but ill give a shot If the credit min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ; SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00] [18/10/2010 07:01:12]:[file:a2billing.php - line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff, activated, inuse, simultaccess, typepaid, creditlimit, language, removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate), expiredays, nbused, UNIX_TIMESTAMP(firstusedate), UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname, cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign, cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON tariff=cc_tariffgroup.id WHERE username='9971524976'] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 :: active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 :: language=en]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD : prepaid-zero-balance (cardnumber:9971524976)]] [18/10/2010 07:01:14]:[file:a2billing.php - line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY : callingcard_ivr_authenticate]] [18/10/2010 07:01:14]:[file:a2billing.php - line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED (cia_res:-2)]] [18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]] -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
[asterisk-users] Is Asterix right tool for me?
Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone equipment would they have to use (VoIP telephones?) Thanks Jane -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best way to recording the hold time for a Queue agent or an extension
Hi Everyone, We are using Queuemetrics but it doesn't Record the Hold Time as it's never logged on the queue_log file. However, when an agent or an extension presses HOLD button on their phone, asterisk does create an event for Music On Hold which is logged in the /var/log/asterisk/full. I want to record the total hold time for an extension and save it with an epoch time stamp. What is the best approach to this? read and parse /var/log/asterisk/full in a cron job every few seconds? Have a presistent PHP-AGI connection to check for hold time events? As much detail as possible on above approaches or other ideas are most appreciated. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterix right tool for me?
In short terms: 1)broadband internet connection 2) Voip phone like a Cisco 7960 3) Sip Trunks from a SIP Trunk provider Thats a short list of what you will need, but you could ditch your local Telcom operator completely, and run VOIP. There are much more knowledgable people about the subject matter than me, but this should at least get you started! Good luck and Welcome to Asterisk! James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:22, jana1...@centrum.cz wrote: Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone equipment would they have to use (VoIP telephones?) Thanks Jane -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterix right tool for me?
i think you can also use softphones installed in your remote offices. regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller [paramedi...@gmail.com] Sent: Wednesday, October 20, 2010 3:34 PM To: exp...@hope.cz; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is Asterix right tool for me? In short terms: 1)broadband internet connection 2) Voip phone like a Cisco 7960 3) Sip Trunks from a SIP Trunk provider Thats a short list of what you will need, but you could ditch your local Telcom operator completely, and run VOIP. There are much more knowledgable people about the subject matter than me, but this should at least get you started! Good luck and Welcome to Asterisk! James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:22, jana1...@centrum.czmailto:jana1...@centrum.cz wrote: Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone equipment would they have to use (VoIP telephones?) Thanks Jane -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterix right tool for me?
Thats right, i completely forgot that option! I run a soft phone on my laptop, which connects back through my verizon wireless aircard to the pbx and allows me to call out while on the go from anywhere! I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:36, GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.com wrote: i think you can also use softphones installed in your remote offices. regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller [ paramedi...@gmail.com] Sent: Wednesday, October 20, 2010 3:34 PM To: exp...@hope.cz; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is Asterix right tool for me? In short terms: 1)broadband internet connection 2) Voip phone like a Cisco 7960 3) Sip Trunks from a SIP Trunk provider Thats a short list of what you will need, but you could ditch your local Telcom operator completely, and run VOIP. There are much more knowledgable people about the subject matter than me, but this should at least get you started! Good luck and Welcome to Asterisk! James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:22, jana1...@centrum.czmailto: jana1...@centrum.cz wrote: Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone equipment would they have to use (VoIP telephones?) Thanks Jane -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] echo on TE122
I have setup a asterisk with freepbx, a TE122 and i have an ISDN. My problem now is that callers are experiencing echo. checked on dmesg i saw this: # dmesg -c dahdi: Disabled echo canceller NLP because of CED rx detected on channel 2 i searched google but found no soolution and i have no idea what that error means. below are other details that might be of value. # dahdi_hardware pci::04:00.0 wcte12xp+d161:8001 Wildcard TE122 # dahdi_scan [1] active=yes alarms=OK description=Wildcard TE122 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE122 location=PCI Bus 04 Slot 01 basechan=1 totchans=31 irq=20 type=digital-E1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI,HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS chan_dahdi.conf contains: echocancel=yes echocancelwhenbridged=yes echotraining=800 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_genconf
Hi, Am Mittwoch, den 20.10.2010, 01:54 -0200 schrieb Flavio Miranda: Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Just one more question, what it means the RED under alarms when I type dahdi show status. It should be OK? the RED-alarm usually means line disconnected or something similar. You should check your wiring or contact Your provider. If You want to use the line, the status should be listed as OK. HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterix right tool for me?
Thanks ALL for reply James, can you please explain a little more what are Sip Trunks and why are Sip Trunks needed? Thanks Jane In short terms: 1)broadband internet connection 2) Voip phone like a Cisco 7960 3) Sip Trunks from a SIP Trunk provider Thats a short list of what you will need, but you could ditch your local Telcom operator completely, and run VOIP. There are much more knowledgable people about the subject matter than me, but this should at least get you started! Good luck and Welcome to Asterisk! James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:22, jana1...@centrum.cz wrote: Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone equipment would they have to use (VoIP telephones?) Thanks Jane -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterix right tool for me?
The simple answer: it takes the digital (VOIP) signals, and connects your calls to the traditional landline network. So if you are in Chicago, and want to call New York City, it will take your call from your office in chicago, route it, and terminate the call to the regular LandLine provider to talk to Joe's Deli in New York City. And the reverse happens. Someone calls your VOIP number, it connects to your SIP Trunk provider, converts it to digital, and sends it to your VOIP system. If i misspoke something, someone please feel free to correct me, but this is my understanding of it. The technical answer, I will have to defer you to someone with more technical knowledge than myself. For example, my SIP Trunk is provided by www.flowroute.com . They seem good thus far. They came at the suggestion of a friend who uses them. They are a prepaid service. You have to put money into an account, and the monthly charges and the cost per call are charged towards that balance. I am in the process of building out FreePBX which uses asterisk for my web hosting business. So far it has worked well. I've tested the call quality on Flowroute and it has been good. Several people i have talked to cant tell im using VOIP. Shop around, find a provider that works for you, and ask for opinions. Good luck, and i hope i made it as clear as mud! :) James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:48, James Miller paramedi...@gmail.com wrote: Thats right, i completely forgot that option! I run a soft phone on my laptop, which connects back through my verizon wireless aircard to the pbx and allows me to call out while on the go from anywhere! I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:36, GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.com wrote: i think you can also use softphones installed in your remote offices. regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller [ paramedi...@gmail.com] Sent: Wednesday, October 20, 2010 3:34 PM To: exp...@hope.cz; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is Asterix right tool for me? In short terms: 1)broadband internet connection 2) Voip phone like a Cisco 7960 3) Sip Trunks from a SIP Trunk provider Thats a short list of what you will need, but you could ditch your local Telcom operator completely, and run VOIP. There are much more knowledgable people about the subject matter than me, but this should at least get you started! Good luck and Welcome to Asterisk! James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Wed, Oct 20, 2010 at 03:22, jana1...@centrum.czmailto: jana1...@centrum.cz wrote: Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone equipment would they have to use (VoIP telephones?) Thanks Jane -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ --
Re: [asterisk-users] Parked calls drop asterisk-1.4.22.1
das sandesh wrote: Can any one share any ideas or opinions? Sandesh, You'll need to create a context called park-dial and then put logic into it on how to handle a call. I have the following in my dial plan: ;*** ;* If the call that was parked, fails to be answered within the 120 seconds ;* rings back to the parking extension and that extension is busy. It will ;* continue to ring until it is answered. If for whatever reason, the ;* call is rejected, it will fail on a no timeout entry in the dial plan ;* This context has been created to send the caller back into the incoming ;* context to keep from dropping the call. ;*** [park-dial] exten = t,1,Goto(office-hours,s,6) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playback in the middle of a call though AMI
Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback in the middle of a call though AMI
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gustavo Garcia Bernardo Sent: Wednesday, October 20, 2010 6:06 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Playback in the middle of a call though AMI Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. I don't use 1.6, but you might be able to do a command/playback to play the file. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI weather quirk
Hello list, This may or may not be Asterisk related, but if I had hair I'd pull it out over this. I have a TDM400P card in a Dell POWEREDGE 1550 running Asterisk 1.4.30. Everything works great except that every time it rains, I get flooded with this CLI message - == Starting post polarity CID detection on channel 1 -- Starting simple switch on 'DAHDI/1-1' [Oct 20 03:17:08] WARNING[8905]: chan_dahdi.c:6827 ss_thread: Channel DAHDI/1-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'DAHDI/1-1' == Starting post polarity CID detection on channel 1 -- Starting simple switch on 'DAHDI/1-1' [Oct 20 03:17:08] WARNING[8906]: chan_dahdi.c:6827 ss_thread: Channel DAHDI/1-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. * Hungup 'DAHDI/1-1' Once it is dry for 12-18 hours, the message goes away. Any suggestions? Thanks Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI weather quirk
The telcom (and even teletype) term is: When it rains, it pours. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Wed, Oct 20, 2010 at 10:33 AM, Danny Nicholas da...@debsinc.com wrote: Hello list, This may or may not be Asterisk related, but if I had hair I’d pull it out over this. I have a TDM400P card in a Dell POWEREDGE 1550 running Asterisk 1.4.30. Everything works great except that every time it rains, I get flooded with this CLI message – == Starting post polarity CID detection on channel 1 -- Starting simple switch on 'DAHDI/1-1' [Oct 20 03:17:08] WARNING[8905]: chan_dahdi.c:6827 ss_thread: Channel DAHDI/1-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'DAHDI/1-1' == Starting post polarity CID detection on channel 1 -- Starting simple switch on 'DAHDI/1-1' [Oct 20 03:17:08] WARNING[8906]: chan_dahdi.c:6827 ss_thread: Channel DAHDI/1-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. n Hungup 'DAHDI/1-1' Once it is dry for 12-18 hours, the message goes away. Any suggestions? Thanks Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Calls Rejection Reasons
Hello all, We would like to inform the caller of the reason for a failed call. For example, when we get a 486 Busy Here, the system accepts it and in the CLI we see Everyone is busy/congested at this time. Can we use this data to play an announcement to the caller? Thank you in advance for your help. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommendation for a new server
Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Calls Rejection Reasons
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VoIP Question Sent: Wednesday, October 20, 2010 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Using Calls Rejection Reasons Hello all, We would like to inform the caller of the reason for a failed call. For example, when we get a 486 Busy Here, the system accepts it and in the CLI we see Everyone is busy/congested at this time. Can we use this data to play an announcement to the caller? Thank you in advance for your help. Michael Yes. You can use the built-in options and sounds like this: Exten = 123,1,dial(SIP/100,30,m) Exten = 123-BUSY,1,playback(all-reps-busy) Exten = 123-BUSY,n,hangup Exten = 123-CONGESTION,1,playback(all-circuits-busy-now) Exten = 123-CONGESTION,n,hangup Or you can get fancy and interpret the return and play back a verbatim message through swift/Cepstral. That's another post. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Calls Rejection Reasons
Yes, look at DIALSTATUS variable that Asterisk set when use DIAL Application. Regards - Bakko-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI weather quirk
On 10/20/2010 09:05 AM, Mark Deneen wrote: On Wed, Oct 20, 2010 at 9:33 AM, Danny Nicholas da...@debsinc.com wrote: Hello list, This may or may not be Asterisk related, but if I had hair I’d pull it out over this. I have a TDM400P card in a Dell POWEREDGE 1550 running Asterisk 1.4.30. Everything works great except that every time it rains, I get flooded with this CLI message – Any suggestions? Talk to your phone company. It sounds like there is water in (on?) your lines. Back when I was on dialup, I had a big problem with this after a rain storm and I could only connect at 9600 baud. +1 for talking to your phone company. I don't believe there is a way the driver / hardware can handle if water is shorting tip and ring on your lines (without breaking something else). -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Thank you Kevin, We'll upgrade our server to 1.6.2.12 and try again. Another question: Is there (expect for the admin guide that we didn't succeed to understand the example in) an example somewhere for ReceiveFax full extensions.conf diaplan? We would like to allocate one of the extensions that our SIP provider gives us to a fax storage server or later to email. Michael On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming kpflem...@digium.comwrote: You have a 'Local' channel in between SendFAX and the SIP channel to your other endpoint. In Asterisk 1.6.2.11, chan_local was not properly aware of T.38 negotiation, so it ends up acting as a sort of 'firewall' between the endpoints. This was fixed in Asterisk 1.6.2.12 and later releases, so if you were running the current version, you wouldn't have experienced this specific problem. This was listed in the ChangeLog for 1.6.2.12, but unfortunately the commit message the developer wrote did not explain why the change was made or what problem it was addressing, so you wouldn't have noticed it. In any case, upgrading to 1.6.2.12 or later will cure this problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Playback randomly stops
We are having issues with asterisk 1.6.2.12-rc1 and 1.6.2.13 with audio playback randomly stopping during calls. A caller goes to voice mail and the prompts stop playing back. IVR prompts stop playing in mid stream. This occurs randomly and is causing quite a problem. I do not see any errors or warring when the playback stops. It has occurred with sip endpoints running both g711 and g729. Any ideas? Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Wed, Oct 20, 2010 at 10:35 AM, VoIP Question voip.quest...@gmail.com wrote: Another question: Is there (expect for the admin guide that we didn't succeed to understand the example in) an example somewhere for ReceiveFax full extensions.conf diaplan? We would like to allocate one of the extensions that our SIP provider gives us to a fax storage server or later to email. Not that I've ever seen. I built mine by reading mailing list archives, then the source for app_fax. +1 for open source. At least one reason such a thing does not exist is that everybody has a different idea of what 'full extensions.conf dialplan' means. In my case, I ReceiveFax, record the call, give it a naming convention that works, convert tiff to pdf, tail a log to a flat txt file, copy the pdf to Winders. That's not going to be the same thing as what a lot of other people want to do. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 step dialing
Hello all, We're trying to build a small IVR application to allow callers to use the Asterisk for outgoing calls in a 2 steps dialing mode. The context for outgoing calls is called outgoing (we have there an LCR and routing mechanism we want to use, depending on the destination). This is what we did, but it doesn't work: exten = _X., 13, Read(ccdest,vm-enter-num-to-call,,,2) exten = _X., 14, NoOp($ccdest) exten = _X., 15, Dial(Local/$ccd...@outgoing,50) The error we get is: chan_local.c:538 local_call: No such extension/context $ccd...@outgoingwhile calling Local channel -- Couldn't call $ccd...@outgoing We know there's a syntax problem in line 15, but not sure how to fix it. Thank you for your help. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] echo on TE122
Thank you Shaun, will try that. will that help on the echo issues users are encountering during calls? On 10/20/10 10:28 PM, Shaun Ruffell wrote: On 10/20/2010 03:20 AM, Ron wrote: I have setup a asterisk with freepbx, a TE122 and i have an ISDN. My problem now is that callers are experiencing echo. checked on dmesg i saw this: # dmesg -c dahdi: Disabled echo canceller NLP because of CED rx detected on channel 2 That's saying that the echo canceller activated for channel 2 is disabled because it believes that the far side is a fax machine or modem. i searched google but found no soolution and i have no idea what that error means. below are other details that might be of value. If you're not actively trying to run faxes through your system, you could try enabling CONFIG_DAHDI_NO_ECHOCAN_DISABLE in include/dahdi/dahdi_config.h. Cheers, Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 step dialing
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VoIP Question
Sent: Wednesday, October 20, 2010 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 2 step dialing
Hello all,
We're trying to build a small IVR application to allow callers to use the
Asterisk for outgoing calls in a 2 steps dialing mode.
The context for outgoing calls is called outgoing (we have there an LCR
and routing mechanism we want to use, depending on the destination).
This is what we did, but it doesn't work:
exten = _X., 13, Read(ccdest,vm-enter-num-to-call,,,2)
exten = _X., 14, NoOp($ccdest)
exten = _X., 15, Dial(Local/$ccd...@outgoing,50)
The error we get is:
chan_local.c:538 local_call: No such extension/context $ccd...@outgoing
while calling Local channel
-- Couldn't call $ccd...@outgoing
We know there's a syntax problem in line 15, but not sure how to fix it.
Thank you for your help.
Michael
Should be (_X.,15,Dial(Local/1/${ccde...@outgoing,50)
Or
_X.,15,Dial(Local/${ccde...@outgoing,50)
${ccdest} returns value of ccdest to dialplan
Local/${ccdest} selects local channel ccdest
You might want local/1 with ccdest value.
Also you might want to look into the DISA command
--
_
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[asterisk-users] SIP 401
Hi I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients with the two accounts it works fine however with Asterisk I am getting SIP 401 In my Sip.conf file I under general register = user:passw...@sip.voipblaster.com then I have a sip peer [FreeCall](default) type= friend context= incoming username = kiks2010 secret = password host= sip.voipblast.com fromuser = kiks2010 fromdomain = sip.voipblast.com insecure=very qualify=yes these are the sip debug logs v=0 o=kiks2010 1287592622 1287592622 IN IP4 77.72.168.99 s=SIP Call c=IN IP4 77.72.168.99 t=0 0 m=audio 11538 RTP/AVP 8 101- --- (11 headers 9 lines) --- == Using SIP RTP CoS mark 5 Sending to 77.72.174.128 : 5060 (NAT) Using INVITE request as basis request - 64de05c42e7b4ef2a0678f999c0ed...@77.72.174.128 Found peer 'FreeCall' for 'ajs2010' from 77.72.174.128:5060 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 --- Reliably Transmitting (NAT) to 77.72.174.128:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 77.72.174.128:5060;branch=z9hG4bK6ff0e241f3fd4d0b9c137d616de1fe1f;received=77.72.174.128 From: ajs2010 sip:ajs2...@sip.voipblast.com:5060;tag=330113ac4c51ef02d4ef70 Any help info will be appreciated thanks Zakir-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 401
Zakir, Have you checked the RFC3261? 21.4.2 401 Unauthorized The request requires user authentication. This response is issued by UASs and registrars, while 407 (Proxy Authentication Required) is used by proxy servers. 2010/10/20 Zakir Mahomedy z...@mayfair2000.com Hi I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients with the two accounts it works fine however with Asterisk I am getting SIP 401 In my Sip.conf file I under general register = user:passw...@sip.voipblaster.comuser%3apassw...@sip.voipblaster.com then I have a sip peer [FreeCall](default) type= friend context= incoming username = kiks2010 secret = password host= sip.voipblast.com fromuser = kiks2010 fromdomain = sip.voipblast.com insecure=very qualify=yes these are the sip debug logs v=0 o=kiks2010 1287592622 1287592622 IN IP4 77.72.168.99 s=SIP Call c=IN IP4 77.72.168.99 t=0 0 m=audio 11538 RTP/AVP 8 101- --- (11 headers 9 lines) --- == Using SIP RTP CoS mark 5 Sending to 77.72.174.128 : 5060 (NAT) Using INVITE request as basis request - 64de05c42e7b4ef2a0678f999c0ed...@77.72.174.128 Found peer 'FreeCall' for 'ajs2010' from 77.72.174.128:5060 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 --- Reliably Transmitting (NAT) to 77.72.174.128:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 77.72.174.128:5060 ;branch=z9hG4bK6ff0e241f3fd4d0b9c137d616de1fe1f;received=77.72.174.128 From: ajs2010 sip:ajs2...@sip.voipblast.com:5060 ;tag=330113ac4c51ef02d4ef70 Any help info will be appreciated thanks Zakir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 401
By the way, Could you please make a better picture of your work? try using insecure=invite,port, that's the key! by the way, try to use IPs rather than domain names. And check here also: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf register = user[:secret[:authuse...@host[:port][/extension] 2010/10/20 Danny Dias ing.diasda...@gmail.com Zakir, Have you checked the RFC3261? 21.4.2 401 Unauthorized The request requires user authentication. This response is issued by UASs and registrars, while 407 (Proxy Authentication Required) is used by proxy servers. 2010/10/20 Zakir Mahomedy z...@mayfair2000.com Hi I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients with the two accounts it works fine however with Asterisk I am getting SIP 401 In my Sip.conf file I under general register = user:passw...@sip.voipblaster.comuser%3apassw...@sip.voipblaster.com then I have a sip peer [FreeCall](default) type= friend context= incoming username = kiks2010 secret = password host= sip.voipblast.com fromuser = kiks2010 fromdomain = sip.voipblast.com insecure=very qualify=yes these are the sip debug logs v=0 o=kiks2010 1287592622 1287592622 IN IP4 77.72.168.99 s=SIP Call c=IN IP4 77.72.168.99 t=0 0 m=audio 11538 RTP/AVP 8 101- --- (11 headers 9 lines) --- == Using SIP RTP CoS mark 5 Sending to 77.72.174.128 : 5060 (NAT) Using INVITE request as basis request - 64de05c42e7b4ef2a0678f999c0ed...@77.72.174.128 Found peer 'FreeCall' for 'ajs2010' from 77.72.174.128:5060 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 --- Reliably Transmitting (NAT) to 77.72.174.128:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 77.72.174.128:5060 ;branch=z9hG4bK6ff0e241f3fd4d0b9c137d616de1fe1f;received=77.72.174.128 From: ajs2010 sip:ajs2...@sip.voipblast.com:5060 ;tag=330113ac4c51ef02d4ef70 Any help info will be appreciated thanks Zakir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterix right tool for me?
Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone equipment would they have to use (VoIP telephones?) Yes, indeed, Asterisk can give you this capability. There are several different approaches which can be used - which one you choose will depend on your needs. You'll need to equip your office users with VoIP telephones. These can be either dedicated IP-capable phones (usually running the SIP voice protocols), or softphone software packages running on their PCs (again, implementing SIP). Dedicated hard IP phones can be had for anywhere from $50 on up. Softphone programs range from completely free to significant amount of money, depending on what capabilities you want. Simple ones will emulate a one-line phone (often with a built-in contact list and autodialer) while more complex ones can emulate a multi-line business phone. You would probably want to equip each PC with a handset or headset of some sort rather than depending on the built- in microphone and speaker. USB-connected handsets are widely available; they're usually marketed as being for Skype, but most of them simply register as USB audio devices and will thus work with almost any soft-phone. You'll want at least one system running Asterisk, to act as the hub for your offices. If you have a large number of users in a particular office, and if they will wish to phone one another within the office or working region, it may make sense to place an Asterisk server in that office so that phone-to-phone traffic stays within the office and doesn't have to travel over the public Internet... this will reduce voice latency (delay) and perhaps reduce your Internet bandwidth costs. Each hard or soft IP phone will register with one of the Asterisk servers, so that it can receive calls through that server. Urgent advice: assign each such phone a unique, difficult-to-guess username (*not* just the extension number you are planning to assign to it) and assign it a *very* difficult- to-guess secret (password). Long, randomly-generated strings of letters, digits, and symbols make the best secrets. You *really* do not want somebody from outside your system to be able to guess a phone's username and password, or they'll be able to make calls overseas for which *you* will be financially responsible (this can be a *very* expensive problem if you don't take care!) As to getting back onto the PSTN (public switched telephone network), there are several different approaches you can take. As others have suggested, the best is probably to purchase a SIP account from one of the many different VoIP providers available. Prices, services, and quality vary. You'll probably be best off picking one which is known to provide good service in your area, and has an Internet-to-PSTN interchange switch close to you (network-wise). This SIP provider can do two things for you: - They can accept outbound SIP calls from your Asterisk server (and/or directly from your IP phones) and route these calls onto the PSTN. This is what you'll want to do, in order to allow your offices that have only Internet connections to make phone calls. - They can provide you with any number of PSTN phone numbers, (in your own country or elsewhere) and route calls to these numbers to your Asterisk server. Phones in your Internet-connected offices could make calls out to the PSTN via any of several methods: - They could place calls directly to the SIP provider's servers. This would have the least latency and overhead, but the worst security problems (every phone would have to have an authorized account with the provider, or share a single outbound account and secret... not a good idea). - They could register with, and then place calls through your organization's main (or only) Asterisk server. The server can restrict call destinations on a per-phone basis if necessary, provide centralized logging, etc. - Offices which have their own Asterisk server, could place calls through that server and out to the SIP provider, rather than going through the main company server. This would provide somewhat better delay and call quality in many cases, and still give you a limited number of somewhat-centralized servers which would manage call security and authorization. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options
[asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hi Everyone, We use the top buttons on Aastra 55i to login and logout from Queues. This is the order: Button 1 = Login to English Queue Button 2 = Login to Spanish Queue Button 3 = Logout of English/Spanish Queues There are indicator LEDs on each of these buttons. Is there anyway we can send a SIP request or some other communication to get the Aastra 6755i phone to keep the LED for login set to ON if agent is logged in and to put it to off when agent logs out? Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Wednesday, October 20, 2010 3:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk? Hi Everyone, We use the top buttons on Aastra 55i to login and logout from Queues. This is the order: Button 1 = Login to English Queue Button 2 = Login to Spanish Queue Button 3 = Logout of English/Spanish Queues There are indicator LEDs on each of these buttons. Is there anyway we can send a SIP request or some other communication to get the Aastra 6755i phone to keep the LED for login set to ON if agent is logged in and to put it to off when agent logs out? Thanks, Bruce Don't know anything about Aastra phones, but this would be a BLF trick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Hello again, If I set a peer to use G.711 only, they try to process a sent fax in G.711, but Asterisk doesn't like it: WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on channel 'SIP/Main-000a' and T.38 negotiation failed; aborting. What can I do to enable it? Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hello, you can't utilice the same butons to know the state of the agent but you can configure the LEDs in the opposite position (4,5,6) in the dialplan just before the command to login to the queue put this line (for english queue): exten = yourextension,n,Set(DEVSTATE(Custom:agenten)=INUSE) for spanish queue exten = yourextension,n,Set(DEVSTATE(Custom:agentes)=INUSE) in the dialplan part relative to agent logoff (english) exten = yourextension,n,Set(DEVSTATE(Custom:agenten)=NOT_INUSE) spanish exten = yourextension,n,Set(DEVSTATE(Custom:agentes)=NOT_INUSE) then on the Aastra 6755i web page (on the Programmable Keys menu): keytypevalueline 4BLFagentenglobal 5BLFagentesglobal Now each time the agent login to english queue the 4 key LED switch to red. The same with key 5 LED Please try and give us a feedback Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adaptive CDR and default fields
I'm running 1.6.2.13 and need to record a small number of custom values use cdr_odbc and cdr_adaptive_odbc, and only the custom fields. The good news is that the custom records are being stored in the database as desired. The bad news is that I get three sets of warnings/notice about 'SQL Exec Direct failed' and dropping then reconnecting the database handle. I traced the SQL calls and found these occur when the CDR engine attempts to record all of standard CDR fields. The cdr_adaptive_odbc documentation suggests that it is safe to drop the standard fields, and while my system does continue to function the dropping of the db handle and extra logging is annoying. Have I missed an option to disable recording the standard CDR fields? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Amazing. Thank you very much. Unfortunately, the phone type is 53i and not the 55i as I mistakenly noted. It has only 6 buttons on the left side. Is there a workaround for this? Thanks again. -Bruce On Wed, Oct 20, 2010 at 5:12 PM, bakko asannu...@gmail.com wrote: Hello, you can't utilice the same butons to know the state of the agent but you can configure the LEDs in the opposite position (4,5,6) in the dialplan just before the command to login to the queue put this line (for english queue): exten = yourextension,n,Set(DEVSTATE(Custom:agenten)=INUSE) for spanish queue exten = yourextension,n,Set(DEVSTATE(Custom:agentes)=INUSE) in the dialplan part relative to agent logoff (english) exten = yourextension,n,Set(DEVSTATE(Custom:agenten)=NOT_INUSE) spanish exten = yourextension,n,Set(DEVSTATE(Custom:agentes)=NOT_INUSE) then on the Aastra 6755i web page (on the Programmable Keyshttp://192.168.100.100/programkey.html menu): keytypevalueline 4BLFagentenglobal 5BLFagentesglobal Now each time the agent login to english queue the 4 key LED switch to red. The same with key 5 LED Please try and give us a feedback Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adaptive CDR and default fields
On Wed, Oct 20, 2010 at 5:56 PM, Dan Austin dan_aus...@phoenix.com wrote: The cdr_adaptive_odbc documentation suggests that it is safe to drop the standard fields, and while my system does continue to function the dropping of the db handle and extra logging is annoying. Have I missed an option to disable recording the standard CDR fields? Have you reloaded the module within asterisk? *CLI module reload cdr -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] echo on TE122
On 10/20/10 11:04 AM, Ron wrote: Thank you Shaun, will try that. will that help on the echo issues users are encountering during calls? If all the echo problems are due to erroneous tones, then I believe it should. If your users are still reporting echo you might want to contact Digium technical support for help with troubleshooting. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Email from Dialplan
Hi, I'm sure this topic has been discussed before but i'm having trouble finding a simple answer. Whats the easiest way of sending an email from Asterisk? I want to set up a warning so that after a Dial cmd, if the DIALSTATUS is CHANUNAVAIL, Asterisk sends an email to the admin to check the voip phone is connected properly. I've got the dial plan set up, I just dont know what command to use to send the email. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hi, you can use 4 for login/logoff (english and spanish) and two for online/offline The procedure is the same. Regards - Bakko-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Email from Dialplan
On Wed, 20 Oct 2010, Dan Journo wrote:
I want to set up a warning so that after a Dial cmd, if the DIALSTATUS is
CHANUNAVAIL, Asterisk sends an email to the
admin to check the voip phone is connected properly.
I've got the dial plan set up, I just dont know what command to use to send
the email.
Off the top of my head...
exten = *,n,system(echo Call to ${DNIS} failed | mail -s 'Call failed'
r...@localhost)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Recommendation for a new server
Any suggestions? On Wed, Oct 20, 2010 at 9:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -- Zeeshan A Zakaria -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
