[Asterisk-Users] Avoiding initial deadlock on iax?
Hi, My asterisk sometimes stop responding to iax calls. In the log, I've found this: Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) - decrement call limit counter Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 WARNING[8386] channel.c: Avoided initial deadlock for '0x81d9530', 10 retries! It happens unpredictably and all I can do just killall -9 asterisk :S. When I execute iax2 show channels on CLI, I got messages that indicate many iax channel hung and I cannot do soft hangup to them :(. Here is my iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) tos=0x68 ; bandwidth=low jitterbuffer=yes dropcount=2 disallow=all allow=ilbc ;allow=g723.1 ;allow=g729 ;allow=ulaw ;allow=alaw ;allow=gsm mailboxdetail=yes the other settings on iax.conf are just iax2 account for trunk and personal use. So I cut them in order to save spaces... Perhaps it's a bug? I've found this http://bugs.digium.com/view.php?id=4045 , but from the link I read that it is just for H323 not for iax. Will that patch cure my asterisk problem since the symptom are the same? Anyone has any ideas? Thanks Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer - (Call failed)
Hi,
I'm trying to call an extension and then transfer the call
to another extension, but something strange happens.
This is the extension:
exten = _9.,1,Dial(CAPI/ISDN4/${EXTEN:1}/b,tT)
When I dial any number starting with 9, I always
get CALL FAILED, but the called party still receive
the call and hears silence forever.
Does anyone know why?
Giuseppe
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Re: [Asterisk-Users] AAH: DNID not set if caller suppresses CID?
Hi, just to complete this thread if someone faces a similiar problem: The missing DID is caused by our telco company. It only happens when having two different ptp lines (with different numberblocks) and calling from one of these to the other. Calls from any other line in the world come in with the DID set to the correct extension. Rgds, Hans ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell 2850 w/TDM2400?
Or just pop down to your local computer store and get a molex splitter. Regards, Mark. Rich Adamson wrote: The fxs ports have to generate ringing voltage (about 90 vac) and they use the 12 volt power supply to do that. When an fxs port is not ringing, it consumes about the same amount of power as an fxo port; not much. The 12 volt power is not available via the pci connector, so the TDM cards use one of the disk drive connectors commonly found in most chassis. The majority of 1U and 2U chassis do not have extra disk drive power connectors wired in them. If you're handy with electronics and a soldier gun, you can find and rig the 12 volt power required for the cards in those chassis. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 2850 w/TDM2400?
That implies that the 2850 has a standard molex connector anywhere inside of it, which is not the case. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Davies Sent: Wednesday, March 29, 2006 6:34 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell 2850 w/TDM2400? Or just pop down to your local computer store and get a molex splitter. Regards, Mark. Rich Adamson wrote: The fxs ports have to generate ringing voltage (about 90 vac) and they use the 12 volt power supply to do that. When an fxs port is not ringing, it consumes about the same amount of power as an fxo port; not much. The 12 volt power is not available via the pci connector, so the TDM cards use one of the disk drive connectors commonly found in most chassis. The majority of 1U and 2U chassis do not have extra disk drive power connectors wired in them. If you're handy with electronics and a soldier gun, you can find and rig the 12 volt power required for the cards in those chassis. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set caller ID for outgoing PRI calls
Dmitry, it seems to me that just in the definition of the extension, (Outbond CID - thru the AMP) you just define the CID of that extension. be carefull to give the proper CID, within your block, to that extension. good luck, Mickey On 3/29/06, Dmitry Ivanov [EMAIL PROTECTED] wrote: On Tuesday 28 March 2006 16:33, Tomislav Vojvodic wrote: Is that what you were asking?My question is: how can I set specific caller id for outgoing PRI calls? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Agent in multiple queues?
You just add the same agent to both queues (don't use groups), like in queues.conf: [queue1] member=Agent/101 [queue2] ... member=Agent/101 Now Agent 101 is a member of both queues, and will not be called while s/he is on conversation. l. On Wed, 29 Mar 2006 09:02:11 +0200, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Just add the agent to both queues, * will take care of the rest. l. I have tried to put agents in groups and then join groups to specific queue. It doesn't work. I don't know is the problem because one agent can't be in more groups or joining groups to queue's doesn't work. Do you know anything about this? -- Tomislav Parcina tparcina#lama.hr -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
We tried to give a MAX4000 behind a Asterisk with TE 405, but the connection is very slow (max of 28.8) and we have also a problem with a Fax Server behind the Asterisk, We loose lines and so on.Did anyonehave an idea?ThanksNico-- -Ursprüngliche Nachricht-Von: Don Pobanz [EMAIL PROTECTED]Gesendet: Tuesday, 28. Mar 2006 19:19 +0200An: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comBetreff: Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405PNico Giefing wrote: Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? I do the v90 dial up. The modem is connected to an Adtran 750 channel bank. Our DID trunks are on a T1 line to the phone company. If you have analog lines to the phone company it will not work since only 1 A/D conversion is allowed! We aren't doing any IDSN. It ?may? be possible. Don Pobanz Nico Giefing --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Agent in multiple queues?
Ok, Understood all this.. but isn't that for making 'static' agents? What if I want my agents to be able to log in/out of the queues... ie when they are not here. On 3/29/06, Lenz [EMAIL PROTECTED] wrote: You just add the same agent to both queues (don't use groups), like in queues.conf: [queue1] member=Agent/101 [queue2] ... member=Agent/101 Now Agent 101 is a member of both queues, and will not be called while s/he is on conversation. l. On Wed, 29 Mar 2006 09:02:11 +0200, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Just add the agent to both queues, * will take care of the rest. l. I have tried to put agents in groups and then join groups to specific queue. It doesn't work. I don't know is the problem because one agent can't be in more groups or joining groups to queue's doesn't work. Do you know anything about this? -- Tomislav Parcina tparcina#lama.hr -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reporting?
Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP frame size location?
On 03/29/06 13:06 Andres said the following: It works perfectly with other values we have tested of 40 and 60. We currently use 60 on all our servers. It cuts down on bandwidth for a G279 call to about 15Kbps. with 60ms packets, is a packet loss or two noticable ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Oneway Audio
Hi all, I did not get this error in Asterisk 1.2.5 release. I am testing on Asterisk SVN-trunk-r15187 to avail the PARKEDAT variable. - I park the call using ParkAndAnnounce - plays moh. - accept the call using ParkedCall The following errors are coming on the console and there is oneway audio - no audio after Music-On-Hold at caller's side. Please advice. I am testing using cisco 7902 phones and using cisco 2800 router. Codec is g711ulaw regards, -- Executing ParkedCall(SIP/192.168.50.2-09cbd610, 366) -- Channel SIP/192.168.50.2-09cbd610 connected to parked call 366Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Agent in multiple queues?
With such a configuration, agents get to be available (on all queues) as soon as they log in and stop to be availbl when they log out. That's why youu use agents instead of, say, SIP/123. The other alternative is to have an agent join each queue dynamically via AddQueueMember() and then log off from each queue dynamically... but this leads to much larger agent control problems. :-) l. On Wed, 29 Mar 2006 14:59:15 +0200, Matt [EMAIL PROTECTED] wrote: Ok, Understood all this.. but isn't that for making 'static' agents? What if I want my agents to be able to log in/out of the queues... ie when they are not here. On 3/29/06, Lenz [EMAIL PROTECTED] wrote: You just add the same agent to both queues (don't use groups), like in queues.conf: [queue1] member=Agent/101 [queue2] ... member=Agent/101 Now Agent 101 is a member of both queues, and will not be called while s/he is on conversation. l. On Wed, 29 Mar 2006 09:02:11 +0200, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Just add the agent to both queues, * will take care of the rest. l. I have tried to put agents in groups and then join groups to specific queue. It doesn't work. I don't know is the problem because one agent can't be in more groups or joining groups to queue's doesn't work. Do you know anything about this? -- Tomislav Parcina tparcina#lama.hr -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS in Spain (it seems Protocol 2)
Hello, (I have asked it some time ago in Asterisk-es mailing list, so excuse me if anybody receive it twice.) I am trying to send SMS in Spain using landlines. It seems that app_sms.c only handles Protocol 1, but Spain and Italy are using Protocol 2. I have been searching in Internet without any results... anybody is sending SMS from Asterisk (or any method) using Protocol 2? (so, it seems, Spain or Italy?) If nobody is able to send, is there more people interested on it? Or any project/person/firm trying to send SMS using Protocol 2? Thank you very much, PD: some guy from Asterisk-es said to me that it seems that Telefonica wants to implement Protocol 1 too... but I don't have any information about deadlines, etc... -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Vonage
I know Vonage doesnt officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receptionist Phones
The reality is, of course, that telephone systems have provided this function for many years. A DSS/BLF is available on MANY so called legacy systems, so until this function is readily available , customers that require a receptionist will continue to go elsewhere. Perhaps it is time to rethink the way data is exchanged between the CPU and the DSS/BLF? As someone said a very long time ago: Results, not excuses. JMO John Novack Christian Stredicke wrote: Well the problem with the sidecar is simple. Just try to light all lights three times within one second. If you have 50 keys there is already hell breaking loose. If you cascade side cars and say have 100 LED, this is a real Xmas tree. The CPU drowns in XML notifications. We already had trouble, and we don't want to double it at this time. Good work, IETF. BTW this is not only a problem if the phone. If the PBX has to supply 50 phones with 50 LED and e.g. they are going off hook at the same time, we are talking about a burst of 50 * 50 = 2500 messages which will have some impact of the PBX CPU as well. We need to do something about this first before we can start having 100 or 150 LED on a device. Christian - yes I am from snom. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mustardman29 Sent: Tuesday, March 28, 2006 8:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Receptionist Phones So how did the Polycom with sidecars work? I like the idea of a dedicated FOP display but not sure why you would need it if you have a Polycom with sidecars. -Original Message- From: Jerry Jones [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 28, 2006 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Receptionist Phones We installed a snom with 3 sidecars. Kinda worked, but had so many quirks they had us replace with a Polycom. All their other phones were of the poly variety. We installed a dedicated lcd running FOP for display. Receptionist was much happier. One of the key problems was she like to set the handset on her desk. But then the snom would not ring. On Mar 28, 2006, at 9:01 AM, Bob McDowell wrote: Can you chain these to get more that 42 buttons? I need about 60... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darrell Long Sent: Monday, March 27, 2006 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Receptionist Phones The 360 has an expansion unit. It adds 42 extensions. Darrell S. Long BestWeb Corporation Daniel Hazelbaker wrote: Hmm, which phone from Snom are you using for this? I've looked around their website and I can only find 3 VoIP phones, the 300, 320 and 360. The 360 by the looks of it only has 12 buttons you can assign to different extensions; am I missing something or is that the phone and you just do 12 per phone? Daniel On Mar 27, 2006, at 2:28 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yes - set up about 10 of them at a business last year. Monitoring is fine - picking up calls is a bit iffy at the best of times. (that is, picking up a ringing call by pushing the extension button. *8 works fine) Paul Hales Technical Manager AsteriskIT ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Re: Agent in multiple queues?
LOL Yes it does. Ok.. so you are saying I should put in agent numbers in the queues rather then phone extensions? I'm having serious problems with the agent joining queue... for some reason asterisk is letting me log agents in that aren't configured in agents.conf and then shows them logged into the queue but show agents shows they aren't logged in. So if I set the AGENT to be static to the queue, then how do I log the agent in without logging into the queue? On 3/29/06, Lenz [EMAIL PROTECTED] wrote: With such a configuration, agents get to be available (on all queues) as soon as they log in and stop to be availbl when they log out. That's why youu use agents instead of, say, SIP/123. The other alternative is to have an agent join each queue dynamically via AddQueueMember() and then log off from each queue dynamically... but this leads to much larger agent control problems. :-) l. On Wed, 29 Mar 2006 14:59:15 +0200, Matt [EMAIL PROTECTED] wrote: Ok, Understood all this.. but isn't that for making 'static' agents? What if I want my agents to be able to log in/out of the queues... ie when they are not here. On 3/29/06, Lenz [EMAIL PROTECTED] wrote: You just add the same agent to both queues (don't use groups), like in queues.conf: [queue1] member=Agent/101 [queue2] ... member=Agent/101 Now Agent 101 is a member of both queues, and will not be called while s/he is on conversation. l. On Wed, 29 Mar 2006 09:02:11 +0200, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Just add the agent to both queues, * will take care of the rest. l. I have tried to put agents in groups and then join groups to specific queue. It doesn't work. I don't know is the problem because one agent can't be in more groups or joining groups to queue's doesn't work. Do you know anything about this? -- Tomislav Parcina tparcina#lama.hr -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Vonage
Brian Deep posted this to the list back in August. I still haven't tried it
myself but he said it worked. If you try it out and it works, please post back
your success. (or failure if that is the case)
Jason
sip.conf:
[general]
externip=X.X.X.X
port=5060
bindaddr=X.X.X.X
context=vonage-out
disallow=all
allow=ulaw
allow=alaw
nat=yes
register=:[EMAIL PROTECTED]:5060/201
[vonage]
username=
type=peer
secret=PASSWORD
port=5060
nat=yes
host=atlas-east.vonage.net
fromdomain=vonage.net
canreinvite=no
fromuser=
dtmfmode=rfc2833
context=vonage-out
[201]
type=friend
username=201
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
defaultip=X.X.X.X
mailbox=201
callerid=NAME
progressinband=no
context=from-sip
extension.conf:
[vonage-out]
exten = ,1,Goto(from-sip,201,1)
[from-sip]
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones
Sent: Wednesday, March 29, 2006 8:59 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk with Vonage
I know Vonage doesn't officially have a bring your own device type program,
but they do offer a softphone. Has anyone gotten Asterisk to connect directly
to Vonage? This would be a great help!!
___
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Re: [Asterisk-Users] Asterisk with Vonage
On 29/03/06, Steve Jones [EMAIL PROTECTED] wrote: I know Vonage doesn't officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! I'm not a Vonage customer, but I did spot this: http://blog.tmcnet.com/blog/tom-keating/vonage/vonage-opens-sip-credentials.asp Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Vonage
Plus see this: http://www.voip-info.org/wiki/view/Asterisk+and+Vonage Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Wednesday, March 29, 2006 8:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with Vonage On 29/03/06, Steve Jones [EMAIL PROTECTED] wrote: I know Vonage doesn't officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! I'm not a Vonage customer, but I did spot this: http://blog.tmcnet.com/blog/tom-keating/vonage/vonage-opens-sip-credenti als.asp Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Vonage ATA question follow-up
On the same line I see we can not connect Asterisk to Vonage. But has anyone tried unlocking the Motorola ATA they send you? I would like to configure it so that I can connect to my office Asterisk installation. Thank you, Jyran Glucky Bob McDowell [EMAIL PROTECTED] lprotection.com To Sent by: [EMAIL PROTECTED] asterisk-users-bo [EMAIL PROTECTED], Asterisk Users [EMAIL PROTECTED] Mailing List - Non-Commercial m.com Discussion asterisk-users@lists.digium.com cc 03/29/2006 09:24 AMSubject RE: [Asterisk-Users] Asterisk with Vonage Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Plus see this: http://www.voip-info.org/wiki/view/Asterisk+and+Vonage Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Wednesday, March 29, 2006 8:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with Vonage On 29/03/06, Steve Jones [EMAIL PROTECTED] wrote: I know Vonage doesn't officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! I'm not a Vonage customer, but I did spot this: http://blog.tmcnet.com/blog/tom-keating/vonage/vonage-opens-sip-credenti als.asp Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Marketing Materials
The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Thanks, Bob McDowell EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Vonage
I have this working. I have Asterisk connecting to my Vonage Linksys device via Digium Wildcard X100P. No magic needed ;) Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: Wednesday, March 29, 2006 9:25 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk with Vonage Plus see this: http://www.voip-info.org/wiki/view/Asterisk+and+Vonage Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Wednesday, March 29, 2006 8:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with Vonage On 29/03/06, Steve Jones [EMAIL PROTECTED] wrote: I know Vonage doesn't officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! I'm not a Vonage customer, but I did spot this: http://blog.tmcnet.com/blog/tom-keating/vonage/vonage-opens-sip-credenti als.asp Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Small - Medium Billing Software needed
On 3/27/06, Erick Perez [EMAIL PROTECTED] wrote: Hi, Our asterisk installation will be a man-in-the-middle providing local,long,international VOIP services to our customers and our asterisk will be connect via VOIP to international carriers. We use asterisk 1.2.5 with mysql in centos 4.2 Kernel 2.6 I have looked at astbill and it sounds interesting, but their forum seems dead (lack of activity or a lot of unanswered questions). Any other suggestions for some open source or commercial billing systems for small installations like mine? We are looking for (this list is not complete) 1.2.5 Asterisk support in realtime h323, gsm and g729 web configuration if possible Dynamic International Rate Table (Each customer can have his own price list using Brands) LCR web based if possible MySQL based (because we use realtime asterisk) prepaid / postpaid (but *not* interested in online credit card processing at this moment) Switchboard (Displays live status of users phones and ongoing calls) this message will be posted to the bussiness list. Thanks to all. -- --- Erick -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX AAH
snip wonderful place to start. Nothing against Asterisk or Linux. My build fromscratch issues are only due to my lack of Linux experience.../snip the only way to learn is by playing. a little over a year ago i knew nothing about linux. google. is your friend. New Yahoo! Messenger with Voice. Call regular phones from your PC and save big.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Marketing Materials
Bob McDowell wrote: The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Bob, Check on Digium's website. I know there is such a creature there. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sipura spa2 + asterisk bug ?
Steve Kennedy wrote: On Tue, Mar 28, 2006 at 07:40:06PM +, Tofik Suleymanov wrote: Each of the two lines have their own entry in sip.conf and i can see each line registered in 'sip show peers'. I can dial each line from outside successfully but when one line is busy i can't reach the second line (it immediately sends me to the voicemail).I've also tried to change the timeouts in dial command but seems that it doesn't matter. Any other advice ? You haven't got codec negotiation set-up properly so it's still running out of g.729 and then it will act as busy I have dtmfmode=rfc2833 disallow=all allow=g729 allow=gsm allow=alaw allow=ulaw allow=g723.1 So should try g.729 first, then gsm (which unfortunately SPA don't support), etc etc. Steve Thank you very much ! after playing a bit with codecs i've managed my sipura lines to work properly. Again, thank you very much for quick and effective help ! Tofik Suleymanov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sipura spa2 + asterisk bug ?
Vahan Yerkanian wrote: Tofik Suleymanov wrote: 1. assume 1-st line is in use 2. after dialing 2-nd line from outside i immediately go to the voicemail announcement (also i immediately go to voicemail if i dial from extension to extension both of which are on the same sipura device) Check what response code appears on Asterisk CLI when you dial 2nd line. If your SPA-2k is SPA-2000 or SPA-2002 there might be a codec combination that is still overloading CPU and it's sending back unavailable response. I assume both extensions have separate username/passwords, don't they? Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you, Vahan i've managed both of lines to work after playing a bit with codecs. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Routing SIP calls via URI
Dear All,
I have the following setup;
SER/External Asterisk -- Firwall -- Internal Asterisk -VPN- Users
At the moment;
Anybody can register with our SER proxy and call each other using VoIP.
Anybody can call one of our internal users via our SER/Asterisk gateway.
The INVITE is sent to our external Asterisk Server, this act as a UA and
uses IAX2 to send the call to our internal Asterisk server.
Our internal users use a VPN to connect to our corporate HQ. They
register with our Internal Asterisk server and can make internal and
PSTN calls.
What I would like to do is to redirect external SIP calls to our
external Asterisk server. e.g if I call sip:[EMAIL PROTECTED] I would
like the call to be routed from our Internal Asterisk server to our
External Asterisk server via IAX2 and for the external asterisk server
to act as a UA and make the call.
I have tried the following syntax on our internal server;
exten = _sip.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
However this does not seem to work?
How do I change my dial plan so that SIP calls are routed from my
internal Asterisk box to my external Asterisk box over IAX2?
Warm Regards and Thanks
Shad Mortazavi
---
Nexus Management Inc
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RE: [Asterisk-Users] Marketing Materials
I did. I will do so again, just to be sure. I am one of those 'search first' type of list guys, and would not want to waste this list's time if I could help it... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darrick Hartman Sent: Wednesday, March 29, 2006 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Marketing Materials Bob McDowell wrote: The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Bob, Check on Digium's website. I know there is such a creature there. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Marketing Materials
Digium.com has pdf brochures on Asterisk and their hardware you can download. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: Wednesday, March 29, 2006 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Marketing Materials The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Thanks, Bob McDowell EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS in Spain (it seems Protocol 2)
Capatres released some time ago a solution with an ITSP. Maybe it could help http://blogs.capatres.com/index.php?op=ViewArticlearticleId=18blogId=1 Carles Pina i Estany escribió: Hello, (I have asked it some time ago in Asterisk-es mailing list, so excuse me if anybody receive it twice.) I am trying to send SMS in Spain using landlines. It seems that app_sms.c only handles Protocol 1, but Spain and Italy are using Protocol 2. I have been searching in Internet without any results... anybody is sending SMS from Asterisk (or any method) using Protocol 2? (so, it seems, Spain or Italy?) If nobody is able to send, is there more people interested on it? Or any project/person/firm trying to send SMS using Protocol 2? Thank you very much, PD: some guy from Asterisk-es said to me that it seems that Telefonica wants to implement Protocol 1 too... but I don't have any information about deadlines, etc... -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 Fax./Fax.: +34 91 661 9460 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inbound routing help
Hi all I am setting up a asterisk iam able to setup a outbound calling its wrking still some tweaking required iam doing now i got some DID routing to my IP from my provider i want them to give to another Client from my network or out side my network using my asterisk i am not looking forward to extention iam looking the user need to configure directly DID and authenticate with my server and make calls and when i get incoming call to asterisk with that DID, the call should go to appropiate user can some one guide me how can i achieve this any examples ram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Vonage
Actually, they do have a bring your own device program. It's called Business Plus. Works great with Asterisk. http://www.thevoipconnection.com/store/catalog/product_16220_Vonage_Business_Plus.html Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Steve Jones [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 29, 2006 8:59 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk with Vonage I know Vonage doesnt officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX - only one way traffic
Hello all!
Ive got a problem with the IAX setup. Im
previously only experienced with SIP, so that may be part of the problem. However,
Ive managed to register with the IAX server without any trouble
(register line apparently works as it should), and I am also ale to make
outbound calls.
However, for inbound calls, all I get is this (from iax2
debug):
Mar 29 17:44:18 NOTICE[11502]: chan_iax2.c:6794 socket_read:
Rejected connect attempt from iax.providers.server.net, who was trying to reach
'{EXTEN}@'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX
Subclass: REJECT
Timestamp: 01015ms SCall: 1 DCall: 2 [iax.providers.server.net:4569]
CAUSE : No authority found
CAUSE CODE : 50
Any help would be appreciated.
Regards,
Bjorn
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database: 268.3.3/295 - Release Date: 28.03.2006
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[Asterisk-Users] FOP flash panel: how to reload config files when running
Hi, is it possible to force FOP to reload its configuration files (op_buttons.cfg and op_style.cfg) while it is working? I tried to click on the refresh icon but nothing happens. TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AAH lost my IVR phrases
Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: Thank you for calling . I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the phpconfig for Asterisk PBX page, and selected the extensions_additional.conf page. On this page were the entries for the Normal and After Hours greetings. The initial greeting phrases were expressed in terms of statements like: exten = s,n,Background(custom/aa_num). It was easy to extend the greeting (for instance, Office hours are 7-7, Press pound for directory..) by directly adding more canned phrases, like so: exten = s, n+1, Background(custom/aa_num+1) etc... Hit update, Re-Read Configs. Try it out. It worked fine. And I felt pretty clever. For a few weeks. Then a complaint: Callers encountered an obviously truncated IVR script, and had no way out of the maze. Sure enough, only one phrase was being uttered. And, sure enough, only one phrase was being commanded by the existing extensions_additional.conf file. I re-edited the file, updated, and things worked again. !!!? What happened to my edited, updated, and Re-Read extensions_additional.conf file? Anybody ever encounter this behavior? What to do, in order to avoid this mishap in the future? Ideas, thoughts? Thanks, Jim Hanlon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TFTP problems on FC4
Hello,
from where do you see errors which are generated by tftp. I have
searched from Google and I didn't find that. Maybe you can help me. I
have FC4 too :)On 3/17/06, Joseph Rothstein [EMAIL PROTECTED] wrote:
Greetings to all.I am hoping someone can help me out with a problem I am having getting myCisco phones, 7960s and 7940s, to download the appropriate files from ourTFTP server. The TFTP server is running on Fedora Core 4.
The TFTP server appears to be setup properly:service tftp{socket_type
= dgramprotocol=
udpwait=
yesuser=
rootserver=
/usr/sbin/in.tftpdserver_args
= - -c -s /tftpboot/disable
= noper_source=
11cps
= 100 2flags
= IPv4}Here is the error message I get on FC4:Mar 17 09:52:52 localhost in.tftpd[22194]: RRQ from 172.16.10.56 filenameOS79XX.TXTMar 17 09:52:52 localhost in.tftpd
[22194]: sending NAK (0, Permissiondenied) to 172.16.10.56Mar 17 09:52:52 localhost in.tftpd[22195]: RRQ from 172.16.10.56 filenameSIPDefault.cnf
Mar 17 09:52:52 localhost in.tftpd[22195]: sending NAK (0, Permissiondenied) to 172.16.10.56Mar 17 09:52:52 localhost in.tftpd[22196]: RRQ from 172.16.10.56
filenameSIP00156316C1CC.cnfMar 17 09:52:52 localhost in.tftpd[22196]: sending NAK (0, Permissiondenied) to 172.16.10.56The phone shows these messages:W200 TFTP Error: Enclosed
Text 0S79XX.txtW210 TFTP Error: EnclosedText SIPDefault.cnfW220 TFTP Error: EnclosedText SIP00156316C1CC.cnfThe phones and the server are in the same VLAN.Has anyone seen this before or have any ideas why simple TFTP does not work?
Regards,Joe___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Agur Koort
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Re: [Asterisk-Users] Receptionist Phones
Because we still have to live with - the current - seven only limitation. We have found that training generally results in folks understanding how to use a PBX vs a key system For many years I installed large PBX systems, Rolms, ATT, Nortel, etc and often no blf/dss was available, or management did not wish to purchase. Users at first were unhappy, but got used to working with the PBX. In fact most had to give up key phones with lots of buttons/ lights and oonly had a stnadard 2500 set to do everything with. On Mar 28, 2006, at 7:33 PM, mustardman29 wrote: So how did the Polycom with sidecars work? I like the idea of a dedicated FOP display but not sure why you would need it if you have a Polycom with sidecars. -Original Message- From: Jerry Jones [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 28, 2006 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Receptionist Phones We installed a snom with 3 sidecars. Kinda worked, but had so many quirks they had us replace with a Polycom. All their other phones were of the poly variety. We installed a dedicated lcd running FOP for display. Receptionist was much happier. One of the key problems was she like to set the handset on her desk. But then the snom would not ring. On Mar 28, 2006, at 9:01 AM, Bob McDowell wrote: Can you chain these to get more that 42 buttons? I need about 60... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darrell Long Sent: Monday, March 27, 2006 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Receptionist Phones The 360 has an expansion unit. It adds 42 extensions. Darrell S. Long BestWeb Corporation Daniel Hazelbaker wrote: Hmm, which phone from Snom are you using for this? I've looked around their website and I can only find 3 VoIP phones, the 300, 320 and 360. The 360 by the looks of it only has 12 buttons you can assign to different extensions; am I missing something or is that the phone and you just do 12 per phone? Daniel On Mar 27, 2006, at 2:28 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yes - set up about 10 of them at a business last year. Monitoring is fine - picking up calls is a bit iffy at the best of times. (that is, picking up a ringing call by pushing the extension button. *8 works fine) Paul Hales Technical Manager AsteriskIT ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SMS in Spain (it seems Protocol 2)
Telefónica use both protocols to deliver an SMS (UBS1 and UBS2). Most nowadays fixed-devices (in Spain) are UBS2 but there are UBS1 too. The Telefónica messaging platform have the information of terminals of each subscriber and its access protocol. good luck, hope it helps!! Fran -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Alberto Sagredo Enviado el: miércoles, 29 de marzo de 2006 17:19 Para: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SMS in Spain (it seems Protocol 2) Capatres released some time ago a solution with an ITSP. Maybe it could help http://blogs.capatres.com/index.php?op=ViewArticlearticleId=18blogId=1 Carles Pina i Estany escribió: Hello, (I have asked it some time ago in Asterisk-es mailing list, so excuse me if anybody receive it twice.) I am trying to send SMS in Spain using landlines. It seems that app_sms.c only handles Protocol 1, but Spain and Italy are using Protocol 2. I have been searching in Internet without any results... anybody is sending SMS from Asterisk (or any method) using Protocol 2? (so, it seems, Spain or Italy?) If nobody is able to send, is there more people interested on it? Or any project/person/firm trying to send SMS using Protocol 2? Thank you very much, PD: some guy from Asterisk-es said to me that it seems that Telefonica wants to implement Protocol 1 too... but I don't have any information about deadlines, etc... -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 Fax./Fax.: +34 91 661 9460 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] IAX - only one way traffic
I should add, I have now managed to get a
little further, but I still get an error message:
Mar 29 18:17:34 NOTICE[11943]:
chan_iax2.c:7213 socket_read: Rejected connect attempt from 213.160.242.5,
request '[EMAIL PROTECTED]' does not exist
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno:
002 Type: IAX Subclass: REJECT
Timestamp: 00036ms SCall: 3
DCall: 2 [213.160.242.5:4569]
CAUSE : No such
context/extension
CAUSE CODE : 3
In my extensions.conf Ive got an entry
for the phone number that Im supposed to receive calls on:
[default]
Exten = 11223344,1,Dial(SIP/1000)
Which basically forwards calls to that SIP
extension. However, why do I receive [EMAIL PROTECTED] from my provider (or
actually, the default part is set in iax.conf) and not the incoming number
instead?
Best regards,
Bjorn
Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] P vegne av Bjorn O
Sendt: 29. mars 2006 17:46
Til: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Emne: [Asterisk-Users] IAX - only
one way traffic
Hello all!
Ive got a problem with the IAX setup. Im previously only
experienced with SIP, so that may be part of the problem. However, Ive managed
to register with the IAX server without any trouble (register line apparently
works as it should), and I am also ale to make outbound calls.
However, for inbound calls, all I get is this (from iax2
debug):
Mar 29 17:44:18 NOTICE[11502]: chan_iax2.c:6794 socket_read:
Rejected connect attempt from iax.providers.server.net, who was trying to reach
'{EXTEN}@'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: REJECT
Timestamp: 01015ms SCall: 1
DCall: 2 [iax.providers.server.net:4569]
CAUSE : No
authority found
CAUSE CODE : 50
Any help would be appreciated.
Regards,
Bjorn
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database: 268.3.3/295 - Release Date: 28.03.2006
--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database: 268.3.3/295 - Release Date: 28.03.2006
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database: 268.3.3/295 - Release Date: 28.03.2006
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Re: [Asterisk-Users] FOP flash panel: how to reload config files when running
is it possible to force FOP to reload its configuration files (op_buttons.cfg and op_style.cfg) while it is working? I tried to click on the refresh icon but nothing happens. No, you have to kill the op_server app and restart it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FOP flash panel: how to reload config files when running
If you are using the safe_opserver daemon, 'killall op_server.pl' works fine. You have to kill the op_server process to force a reread of the .cfg files -Original Message- From: Giorgio Incantalupo [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 29, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] FOP flash panel: how to reload config files when running Hi, is it possible to force FOP to reload its configuration files (op_buttons.cfg and op_style.cfg) while it is working? I tried to click on the refresh icon but nothing happens. TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AAH lost my IVR phrases
You made some change to something using AMP and it overwrote the extensions_additional.conf file as it was designed to do. The only safe place to put customizations is in extensions_custom.conf. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Hanlon Sent: Wednesday, March 29, 2006 8:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AAH lost my IVR phrases Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: Thank you for calling . I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the phpconfig for Asterisk PBX page, and selected the extensions_additional.conf page. On this page were the entries for the Normal and After Hours greetings. The initial greeting phrases were expressed in terms of statements like: exten = s,n,Background(custom/aa_num). It was easy to extend the greeting (for instance, Office hours are 7-7, Press pound for directory..) by directly adding more canned phrases, like so: exten = s, n+1, Background(custom/aa_num+1) etc... Hit update, Re-Read Configs. Try it out. It worked fine. And I felt pretty clever. For a few weeks. Then a complaint: Callers encountered an obviously truncated IVR script, and had no way out of the maze. Sure enough, only one phrase was being uttered. And, sure enough, only one phrase was being commanded by the existing extensions_additional.conf file. I re-edited the file, updated, and things worked again. !!!? What happened to my edited, updated, and Re-Read extensions_additional.conf file? Anybody ever encounter this behavior? What to do, in order to avoid this mishap in the future? Ideas, thoughts? Thanks, Jim Hanlon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_odbc appears to have fields missing
I'm currently using Asterisk running version 1.2.5 and trying to use cdr_odbc to connect to a Microsoft SQL database. I have everything running, but the insert statement being sent to database doesn't appear to have the start, answer, end information in it. Below is the insert statement that MS Profiler shows being sent. As you can see those fields are missing. INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES (@P1,@P2,@P3,@P4,@P5,@P6,@P7,@P8,@P9,@P10,@P11,@P12,@P13,@P14) Here is the record that shows up in cdr-csv ,4718,2599576,default,Asterisk2 4718,SIP/4718-af52,IAX2/visioniax-1,Dial, IAX2/visioniax/[EMAIL PROTECTED],2006-03-29 10:29:23,2006-03-29 10:29:25,2006-03-29 10:29:34,11,9,ANSWERED,DOCUMENTATION that record looks fine. Please let me know if I'm missing anything here. Thanks, -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TFTP problems on FC4
On Wed, 29 Mar 2006 11:05:05 -0500, Agur Koort [EMAIL PROTECTED] wrote:
Hello,
from where do you see errors which are generated by tftp. I have searched
from Google and I didn't find that. Maybe you can help me. I have FC4
too :)
You do a tail on /var/log/messages
By the way other dude, here is my /working/ tftp config. Hope it helps.
/etc/xinet.d/tftp:
service tftp
{
socket_type = dgram
protocol= udp
wait= yes
user= root
server = /usr/sbin/in.tftpd
server_args = -s -v /tftpboot
disable = no
per_source = 11
cps = 100 2
flags = IPv4
log_type= SYSLOG daemon
}
and the ftp folder:
[EMAIL PROTECTED] asterisk]# ls -l /tftpboot/
total 3208
-rw-r--r-- 1 root root350016 Jan 20 12:55 randomfirmware.bin
--
Justin Tunney
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[Asterisk-Users] Installing Cisco IP phone 7910
Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State === == telefon -- SEP00119341E684 None Problem is that my 7910 doesn't stop config. It goes like this: Configuring IP Configuring CM list Opening tftpserveradress and then again ... My sccp.conf [general] servername = Asterisk ; show this name on the device registration keepalive = 30 ; phone keep alive message evey 60 secs. Used to check the voicemail debug = 1 ; console debug level. 1 = 10 context = sccp dateFormat = D.M.Y ; M-D-Y in any order. Use M/D/YA (for 12h format) bindaddr = 192.168.1.24 ; replace with the ip address of the asterisk server (RTP important param) port = 2000 ; listen on port 2000 (Skinny, default) disallow=all; First disallow all codecs allow=alaw ; Allow codecs in order of preference allow=ulaw ; firstdigittimeout = 16 ; dialing timeout for the 1st digit digittimeout = 8; more digits ;digittimeoutchar = # ; you can force the channel to dial with this char in the dialing state autoanswer_ring_time = 1 ; ringing time in seconds for the autoanswer, the default is 0 autoanswer_tone = 0x32 ; autoanswer confirmation tone. For a complete list of tones: grep SKINNY_TONE sccp_protocol.h ; not all the tones can be played in a connected state, so you have to try. remotehangup_tone = 0x32 ; passive hangup notification. 0 for none transfer_tone = 0 ; confirmation tone on transfer. Works only between SCCP devices callwaiting_tone = 0x2d ; sets to 0 to disable the callwaiting tone musicclass=default ; Sets the default music on hold class language=en ; Default language setting ;accountcode=skinny ; accountcode to ease billing deny=0.0.0.0/0.0.0.0 ; Deny every address except for the only one allowed. permit=192.168.1.0/255.255.255.0 ; Accept class C 192.168.1.0 [devices] type = 7910 ; device type (see below) autologin = 30, ; lines list. You can add an empty line for an empty button (7960, 7970, 7940, 7920) description = jj7910 ; internal description. Not important tzoffset = -9 transfer = on ; enable or disable the transfer capability. It does remove the transfer softkey park = on ; take a look to the compile howto. Park stuff is not compiled by default speeddial = ; you can add an empty speedial if you want an empty button (7960, 7970, 7920) speeddial = *97,voicemail, cfwdall = off ; activate the callforward stuff and softkeys cfwdbusy = off dtmfmode = inband ; inband or outofband. outofband is the native cisco dtmf tone play. ; Some phone model does not play dtmf tones while connected (bug?), so the default is inband imageversion = P00405000700 ; useful to upgrade old firmwares (the ones that do not load *.xml from the tftp server) deny=0.0.0.0/0.0.0.0 ; Same as general permit=192.168.1.0/255.255.255.0 ; This device can register only using this ip address dnd = on ; turn on the dnd softkey for this device. Valid values are off, on (busy signal), reject (busy signal), silent (ringer = silent) trustphoneip = no ; The phone has a ip address. It could be private, so if the phone is behind NAT ; we don't have to trust the phone ip address, but the ip address of the connection ;earlyrtp = none ; valid options: none, offhook, dial, ringout. default is none. ; The audio strem will be open in the progress and connected state. private = on ; permit the private function softkey for this device mwilamp = on ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink mwioncall = off ; Set the MWI on call. device = SEP00119341E684 ; device name SEPMAC [lines] id = 30 ; future use pin = 1234 ; future use label = 30 ; button line label (7960, 7970, 7940, 7920) description = Line 30 ; top diplay description context = from-internal ; sccp incominglimit = 2 ; more than 1 incoming call = call waiting. transfer = on ; per line transfer capability. on, off, 1, 0 mailbox = 30 ; voicemail.conf (syntax: [EMAIL PROTECTED]:folder]) vmnum = *97 ; speeddial for voicemail administration, just a number to dial cid_name = JJJ ; caller id name cid_num = 30 trnsfvm = 1000 ; extension to redirect the caller (e.g for voicemail) secondary_dialtone_digits = 9 ; digits for the secondary dialtone (max 9 digits) secondary_dialtone_tone = 0x21 ; outside dialtone musicclass=default ; Sets the default music on hold class language=en ; Default language setting ;accountcode=79501 ; accountcode to ease billing rtptos = 184 ; sets the the rtp packets TOS for this line echocancel = on ; sets the phone echocancel for this line silencesuppression = off ; sets the silence suppression for this line ;callgroup=1,3-4 ; We are in caller groups 1,3,4. Valid for this line ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5. Valid for this
RE: [Asterisk-Users] AstCC
Using this same method would I be able to add a cutsom menu in astcc (like call recording), by having it drop back into the IVR and then back to the agi? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JP Carballo Sent: Wednesday, March 29, 2006 1:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AstCC Il Neofita wrote: Hi, I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes. Any idea how to do? Thank you Yep, one way is to ask for the account code from the dialplan, save it to a var like CARDNO and pass that to astcc.agi When the person is done with a call, they can press, say, *, exit out to the menu, dial 1 and be prompted for a new number to call. The other way is to modify astcc.agi to save the account code to a var CARDNO. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] indications.conf.sample
Just noted that in indications.conf.sample in SVN that in the section [fr] the variable callwaiting is written as callwait. Could someone add the ing? Thanks. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Set caller ID for outgoing PRI calls
My question is: how can I set specific caller id for outgoing PRI calls? Here in Italy I have a E1 PRI line with DID: +39 local-zone-prefix did-block-prefixdid-block-ext I was able to set CallerIDnum only after some attempts: I had to set it only to: did-block-prefixdid-block-ext without using local-zone-prefix, peraphs because I'm connected to an old-style telco central. I set it by: Set(CALLERID(number)=did-block-prefixdid-block-ext) and I was able to debug this using: pri debug span 1 and looking for Presentation: Presentation permitted, user number passed network screening... Bye -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AAH lost my IVR phrases
AAH uses a database to store the configs. It then outputs the database info into the text files for asterisk to use. ALL _additional files are built from the database and hand edits WILL be lost. If you are trying to do something that can't be done in the web interface, you need to put that into the _custom files. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Jim Hanlon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: Thank you for calling . I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the phpconfig for Asterisk PBX page, and selected the extensions_additional.conf page. On this page were the entries for the Normal and After Hours greetings. The initial greeting phrases were expressed in terms of statements like: exten = s,n,Background(custom/aa_num). It was easy to extend the greeting (for instance, Office hours are 7-7, Press pound for directory..) by directly adding more canned phrases, like so: exten = s, n+1, Background(custom/aa_num+1) etc... Hit update, Re-Read Configs. Try it out. It worked fine. And I felt pretty clever. For a few weeks. Then a complaint: Callers encountered an obviously truncated IVR script, and had no way out of the maze. Sure enough, only one phrase was being uttered. And, sure enough, only one phrase was being commanded by the existing extensions_additional.conf file. I re-edited the file, updated, and things worked again. !!!? What happened to my edited, updated, and Re-Read extensions_additional.conf file? Anybody ever encounter this behavior? What to do, in order to avoid this mishap in the future? Ideas, thoughts? Thanks, Jim Hanlon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SJphone Do not send silence - option ? Should be disabled for Asterisk
Hi all, I would like to hear from you, SjPhone has the option to Do not Send silence (audio options, advanced), should i use this or remove this option? Everything ran well until now, but there was few people on my server, i'm increasing sip extensions and i want to avoid complains from the users:) Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: HOWTO: Query channel state on an Ateus Voice Blue GSM gateway
Because the VoiceBlue is only 4 channels and I am supporting 100 cell users,
I needed a way to overflow calls to the PRI of all 4 channels are full.
Unfortunately, there seems to be no built-in mechanism to determine if the
gateway is full, so this script parses the output of asterisk -rx sip show
channels XXX.XXX.XXX.XXX to determine the number of channels currently in
use. Hope this helps someone.
Notes:
1. Execute from /var/lib/asterisk/agi-bin as 'querygsmgateway.agi'
2. Replace .201 with the last dotted quad of your gateway's IP address (in
this example, 192.168.1.201 is the IP of the gateway)
3. Assumes that the variable ${DIALSTRING} is set with the NAMA number of
the mobile to call
querygamgateway.agi:
#!/bin/bash
GSMCHANNEL=`asterisk -rx SIP SHOW CHANNELS | grep -a -A0 .201`
CURRENTCHANNEL1=${GSMCHANNEL:55:7}
CURRENTCHANNEL2=${GSMCHANNEL:118:7}
CURRENTCHANNEL3=${GSMCHANNEL:181:7}
CURRENTCHANNEL4=${GSMCHANNEL:244:7}
TOTALCHANNELS=0
if [ ${GSMCHANNEL:55:7} != ]
then
if [ $CURRENTCHANNEL1 = ulaw ]
then
TOTALCHANNELS=$((TOTALCHANNELS+1))
fi
fi
if [ ${GSMCHANNEL:118:7} != ]
then
if [ $CURRENTCHANNEL2 = ulaw ]
then
TOTALCHANNELS=$((TOTALCHANNELS+1))
fi
fi
if [ ${GSMCHANNEL:181:7} != ]
then
if [ $CURRENTCHANNEL3 = ulaw ]
then
TOTALCHANNELS=$((TOTALCHANNELS+1))
fi
fi
if [ ${GSMCHANNEL:244:7} != ]
then
if [ $CURRENTCHANNEL4 = ulaw ]
then
TOTALCHANNELS=$((TOTALCHANNELS+1))
fi
fi
echo SET VARIABLE GSMCHANNELS \$TOTALCHANNELS\
extensions.conf:
exten = dial,1,AGI(/var/lib/asterisk/agi-bin/querygsmgateway.agi) 'Query
the gateway
exten = dial,3,Gotoif($[$[${DIALSTRING:7:3} = 902]]?4:20) 'Determine if
the dialled # is a mobile. If so, use the gateway
exten = dial,4,Gotoif($[${GSMCHANNELS} = 4]?20:7)'If the gateway is
currently full, dial using the PRI, otherwise
exten = dial,5,ChanIsAvail(SIP/${SECONDARYDIALSTRING:7:[EMAIL PROTECTED])
'If the gateway is down, dial the PRI
exten = dial,6,Dial(SIP/${DIALSTRING:7:[EMAIL PROTECTED],25,Tr) 'Dial using
the gateway
exten = dial,7,InsertVoicemailHandlerHere
exten = dial,20,Dial(ZAP/g0/${DIALSTRING:7:7},25,Tr) 'Dial using the PRI
exten = dial,21,InsertVoicemailHandlerHere
exten = dial,106,Dial(ZAP/g0/${DIALSTRING:7:7},25,Tr) 'Dial using the PRI
if the gateway is down
exten = dial,107,InsertVoicemailHandlerHere
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Re: [Asterisk-Users] FOP flash panel: how to reload config files when running
Giorgio Incantalupo wrote: Hi, is it possible to force FOP to reload its configuration files (op_buttons.cfg and op_style.cfg) while it is working? I tried to click on the refresh icon but nothing happens. killall -HUP op_server.pl 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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FOP flash panel: how to reload config files when running
Wilson Pickett wrote: is it possible to force FOP to reload its configuration files (op_buttons.cfg and op_style.cfg) while it is working? I tried to click on the refresh icon but nothing happens. No, you have to kill the op_server app and restart it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This is incorrect. You can just send it the HUP (Hangup) signal and it will reload it's configuration files. 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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] IAX - only one way traffic
This is really the day for new experiences
sorry for the load on the mailing list, but this will be the last issue
I try to solve before I take the night off;)
So Ive got the incoming calls to
work with a not-so-well solution (could therefore still need some feedback on
the previous email). But now it appears that the IVR menu doesnt respond
to DTMF sent from my cellphone into Asterisk. I did a iax2 debug
and theres no particular output about dtmf keys when I press them on the
cellphone, which basically tells me that the DTMF is probably passed on as
audio from my provider. However, IAX would be pretty useless if theres
no way of accessing the selections in the IVR from a PSTN line.
Ive been searching the list, but
have yet to find a solution to this. Thanks for all help!
Regards,
Bjorn
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Bjørn O
Sendt: 29. mars 2006 18:19
Til: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Emne: SV: [Asterisk-Users] IAX -
only one way traffic
I should add, I have now managed to get a
little further, but I still get an error message:
Mar 29 18:17:34 NOTICE[11943]:
chan_iax2.c:7213 socket_read: Rejected connect attempt from 213.160.242.5, request
'[EMAIL PROTECTED]' does not exist
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno:
002 Type: IAX Subclass: REJECT
Timestamp: 00036ms
SCall: 3 DCall: 2 [213.160.242.5:4569]
CAUSE : No such
context/extension
CAUSE
CODE : 3
In my extensions.conf Ive got an
entry for the phone number that Im supposed to receive calls on:
[default]
Exten = 11223344,1,Dial(SIP/1000)
Which basically forwards calls to that SIP
extension. However, why do I receive [EMAIL PROTECTED] from my provider (or
actually, the default part is set in iax.conf) and not the
incoming number instead?
Best regards,
Bjorn
Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Bjorn O
Sendt: 29. mars 2006 17:46
Til: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Emne: [Asterisk-Users] IAX - only
one way traffic
Hello all!
Ive got a problem with the IAX setup. Im
previously only experienced with SIP, so that may be part of the problem.
However, Ive managed to register with the IAX server without any trouble
(register line apparently works as it should), and I am also ale to make
outbound calls.
However, for inbound calls, all I get is this (from iax2
debug):
Mar 29 17:44:18 NOTICE[11502]: chan_iax2.c:6794 socket_read:
Rejected connect attempt from iax.providers.server.net, who was trying to reach
'{EXTEN}@'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: REJECT
Timestamp: 01015ms SCall: 1
DCall: 2 [iax.providers.server.net:4569]
CAUSE : No
authority found
CAUSE CODE : 50
Any help would be appreciated.
Regards,
Bjorn
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[Asterisk-Users] zaphfc on an 'actual' asterisk?
Hi all I don't manage to get asterisk 1.2.5 or 1.2.6 running with the zaphfc driver The scripts from junghanns.net do download a very old libpri and asterisk version which is too buggy for me to use. Isn't there an acutal patch to get zaphfc support in *? -Benoit- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling home while on the road, will it work?
I have a Digium TDM400P card with 1 FXS and 1 FXO module running on my FreeBSD 6.0 server. While I am on the road, I would like to save on costs by using a soft-phone from my laptop to call in to a telephone connected to this card. I installed both Asterisk and Zaptel drivers from the ports, but still haven't done anything with the configuration files. What else do I require, and what is the mimimum amount of work to get this up and running? Thanks a lot in adavnce. -- Kiffin Rex Gish Gouda, The Netherlands ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc on an 'actual' asterisk?
On 3/29/06, Benoit Panizzon [EMAIL PROTECTED] wrote: Isn't there an acutal patch to get zaphfc support in *? You even have 3 possible ways out.. 1; you stay with the current bristuff (a somewhat older zaptel+asterisk, but is this really making a difference?) 2; you use a visdn snapshot (www.visdn.org) 3; you use mISDN (more info on beronet) Howver, zaptel has the most 'advanced' echo cancellation, so be sure to test it out! If you encounter any pro/contra's, don't hesitate to report back.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstCC
Jeremy wrote: Using this same method would I be able to add a cutsom menu in astcc (like call recording), by having it drop back into the IVR and then back to the agi? Of course. It's just a matter of setting things up in the Dialplan before passing the caller to astcc. In my IVR for instance, option 1 is for making calls, option 2 is for checking balance and option 5 is for checking rates for a particular number. When caller presses 1, he is first prompted for the account number and then passed to astcc.agi, which just asked for the number to call. The caller presses * anytime to exit, and is dropped back into the IVR. If he presses 2, his balance is read back immediately since his account number is already in a variable. If he presses 1 again, astcc asks for another number to call. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 behind a Firewall
There is a proble to put an H323 Asterisk server behind an iptables firewall? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS in Spain (it seems Protocol 2)
Are both protocols enabled? I remember I had to first send an SMS with the Domo (an analog phone with sms capabilities) before I could even receive them. Maybe protocol 1, even if it's implemented, needs to be enabled someway. Julian J. M. On 3/29/06, Fran [EMAIL PROTECTED] wrote: Telefónica use both protocols to deliver an SMS (UBS1 and UBS2). Most nowadays fixed-devices (in Spain) are UBS2 but there are UBS1 too. The Telefónica messaging platform have the information of terminals of each subscriber and its access protocol. good luck, hope it helps!! Fran -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Alberto Sagredo Enviado el: miércoles, 29 de marzo de 2006 17:19 Para: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SMS in Spain (it seems Protocol 2) Capatres released some time ago a solution with an ITSP. Maybe it could help http://blogs.capatres.com/index.php?op=ViewArticlearticleId=18blogId=1 Carles Pina i Estany escribió: Hello, (I have asked it some time ago in Asterisk-es mailing list, so excuse me if anybody receive it twice.) I am trying to send SMS in Spain using landlines. It seems that app_sms.c only handles Protocol 1, but Spain and Italy are using Protocol 2. I have been searching in Internet without any results... anybody is sending SMS from Asterisk (or any method) using Protocol 2? (so, it seems, Spain or Italy?) If nobody is able to send, is there more people interested on it? Or any project/person/firm trying to send SMS using Protocol 2? Thank you very much, PD: some guy from Asterisk-es said to me that it seems that Telefonica wants to implement Protocol 1 too... but I don't have any information about deadlines, etc... -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 Fax./Fax.: +34 91 661 9460 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 behind a Firewall
If you open h323 port and rtp ports, it should work. Il Neofita escribió: There is a proble to put an H323 Asterisk server behind an iptables firewall? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inter-Asterisk Using SIP
I am switching from IAX2 to SIP for my inter-Asterisk transport due to assorted quality issues following the 1.2.4 upgrade. On the server that SENDS the call, I have the following in SIP.CONF: [192.168.1.2_OB] type=peer fromuser=OB host=192.168.1.2 And in EXTENSIONS.CONF exten = 91NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) On the RECEIVING Server in SIP.CONF: [OB] type=user context=longdistance I am not using a REGISTER statement on the receiving server. My problem is that the only way I can seem to get the call delivered into the proper SIP context on the receiving box is to use the fromuser=OB on the sending machine. I tried using username=OB, but then it delivers into the default context. I don't want to use fromuser because it overrides the callerid. Any suggestions? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Routing SIP calls via URI
Shad Mortazavi wrote:
What I would like to do is to redirect external SIP calls to our
external Asterisk server. e.g if I call sip:[EMAIL PROTECTED] I would
like the call to be routed from our Internal Asterisk server to our
External Asterisk server via IAX2 and for the external asterisk server
to act as a UA and make the call.
I have tried the following syntax on our internal server;
exten = _sip.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
However this does not seem to work?
Have you tried this?
exten = shad,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
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[Asterisk-Users] Regulatory Ruling about Caller-ID
Hi, Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? IE... If your name is Joe Smith you can't have Mary Smith set as the caller-id name, unless mary smith is also on your account. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Routing SIP calls via URI
I believe that they covered this exact procedures at www.voip-info.org.
Look for the topic on connecting two Asterisk servers. They outline three
different ways that you can do so.
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Date: Wed, 29 Mar 2006 13:18:07 -0600
Shad Mortazavi wrote:
What I would like to do is to redirect external SIP calls to our
external Asterisk server. e.g if I call sip:[EMAIL PROTECTED] I would
like the call to be routed from our Internal Asterisk server to our
External Asterisk server via IAX2 and for the external asterisk server
to act as a UA and make the call.
I have tried the following syntax on our internal server;
exten = _sip.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
However this does not seem to work?
Have you tried this?
exten = shad,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
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Re: [Asterisk-Users] Calling home while on the road, will it work?
Yes, it will work. On 3/29/06, Kiffin Gish [EMAIL PROTECTED] wrote: I have a Digium TDM400P card with 1 FXS and 1 FXO module running on myFreeBSD 6.0 server.While I am on the road, I would like to save on costs by using a soft-phone from my laptop to call in to a telephone connected to this card.I installed both Asterisk and Zaptel drivers from the ports, but stillhaven't done anything with the configuration files.What else do I require, and what is the mimimum amount of work to get this up and running?Thanks a lot in adavnce.--Kiffin Rex GishGouda, The Netherlands___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
DId you hear about the law that says you can't drive over the posted speed? How about the one about junk faxes, or the one about spam? Seriously, I'd like to see how this will be enforced. While I am sure this is to combat telemarketers faking their caller ID, I'm sure they already have some legal loophole to get through. Kinda like the Do Not Call List that doesn't work, fortunately Asterisk handles that very well. As you can tell, I'm not too keen on new laws, when the old ones don't seem to work either. On 3/29/06, Matt [EMAIL PROTECTED] wrote: Hi,Did anyone hear about a recent ruling which makes it illegal to havecaller-id set to anything except what is on the account of the user? IE... If your name is Joe Smith you can't have Mary Smith set asthe caller-id name, unless mary smith is also on your account.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: AAH lost my IVR phrases
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Sent: Wednesday, March 29, 2006 10:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: AAH lost my IVR phrases AAH uses a database to store the configs. It then outputs the database info into the text files for asterisk to use. ALL _additional files are built from the database and hand edits WILL be lost. If you are trying to do something that can't be done in the web interface, you need to put that into the _custom files. -- Thanks for your comments, much appreciated. It's obvious that Asterisk backs its config info in a database. And it's a great idea. The points I feel are confusing are: 1. The alterations to the config files made via AMP Setup pages are archived in the Asterisk DBMS, but changes made via the AMP Maintenance pages are not (Apparently. It's hard to be sure what the rules are). Such differences in behavior are arbitrary, and quite confusing to a novice Asterisk administrator. 2. The changes made via the Maintenance pages work, at least for a while. And then, after some time passes, or some event occurs (which is it?), the archived entries simply overwrite the existing ones. No warning, no announcement. And no backup. This behavior again is arbitrary; I would much rather have a proposed change simply not work at all, rather than have it work for a time and then suddenly stop. In the former case, I can just keep experimenting; in the latter, I button things up with a sense of accomplishment, only to be chagrined a few days or weeks later. Please don't misinterpret my remarks. Asterisk is a great system, and AAH is a marvel. I have learned a lot in the past couple of months. But I got surprised by Asterisk's backup and restore policies for its config files. Appreciate the advice to put persistent data in _custom files. Jim H. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 behind a Firewall
The h323 channels doesn't have any support for NAT. You'd need to register with a properly configured gnugk for that. Julian J. M. On 3/29/06, Alberto Sagredo [EMAIL PROTECTED] wrote: If you open h323 port and rtp ports, it should work. Il Neofita escribió: There is a proble to put an H323 Asterisk server behind an iptables firewall? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP frame size location?
Dinesh Nair wrote: On 03/29/06 13:06 Andres said the following: It works perfectly with other values we have tested of 40 and 60. We currently use 60 on all our servers. It cuts down on bandwidth for a G279 call to about 15Kbps. with 60ms packets, is a packet loss or two noticable ? Depends on the UA. The ones we use are all Sipura/Linksys. We have measured up to 6% packet loss and the call still sounds pretty good (cell phone quality). A 1-2% packet loss will not degrade the call at all. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: HOWTO: Query channel state on an Ateus Voice Blue GSM gateway
On 10:08, Wed 29 Mar 06, Colin Anderson wrote: Because the VoiceBlue is only 4 channels and I am supporting 100 cell users, I needed a way to overflow calls to the PRI of all 4 channels are full. Unfortunately, there seems to be no built-in mechanism to determine if the gateway is full, so this script parses the output of asterisk -rx sip show channels XXX.XXX.XXX.XXX to determine the number of channels currently in use. Hope this helps someone. We use SetGroup and CheckGroup for this (I know, I should upgrade to 1.2 dialplan functions) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
On Wed, 29 Mar 2006 14:19:41 -0500, Matt [EMAIL PROTECTED] wrote: Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? Where did you hear this? Can you give a link? Looks like I'm going to jail, tee hee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: AAH lost my IVR phrases
This is not an issue with asterisk. Asterisk does not use the database, nor has a web interface. AMP (included with AAH) is an addon that uses a database for it's configs. The only way that asterisk can use them, is to write them out to standard asterisk config files. The rules with AMP, (which I also learned the hard way) is that if it says _additional, do not hand edit it. If it says _ custom.conf, you are safe editing it. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Jim Hanlon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Sent: Wednesday, March 29, 2006 10:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: AAH lost my IVR phrases AAH uses a database to store the configs. It then outputs the database info into the text files for asterisk to use. ALL _additional files are built from the database and hand edits WILL be lost. If you are trying to do something that can't be done in the web interface, you need to put that into the _custom files. -- Thanks for your comments, much appreciated. It's obvious that Asterisk backs its config info in a database. And it's a great idea. The points I feel are confusing are: 1. The alterations to the config files made via AMP Setup pages are archived in the Asterisk DBMS, but changes made via the AMP Maintenance pages are not (Apparently. It's hard to be sure what the rules are). Such differences in behavior are arbitrary, and quite confusing to a novice Asterisk administrator. 2. The changes made via the Maintenance pages work, at least for a while. And then, after some time passes, or some event occurs (which is it?), the archived entries simply overwrite the existing ones. No warning, no announcement. And no backup. This behavior again is arbitrary; I would much rather have a proposed change simply not work at all, rather than have it work for a time and then suddenly stop. In the former case, I can just keep experimenting; in the latter, I button things up with a sense of accomplishment, only to be chagrined a few days or weeks later. Please don't misinterpret my remarks. Asterisk is a great system, and AAH is a marvel. I have learned a lot in the past couple of months. But I got surprised by Asterisk's backup and restore policies for its config files. Appreciate the advice to put persistent data in _custom files. Jim H. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
On 29/03/06, Matt [EMAIL PROTECTED] wrote: Hi, Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? A ruling in what jurisdiction? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
I was told it was more along the lines of preventing stalking, but I'm sure telemarketers fit in there :) IE... Mary Smith seperates from Joe Smith... Joe Smith gets phone service and wants his CID to read 'Jane Smith', who is Mary's sister, so that Mary will answer when Joe calls. On 3/29/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: DId you hear about the law that says you can't drive over the posted speed? How about the one about junk faxes, or the one about spam? Seriously, I'd like to see how this will be enforced. While I am sure this is to combat telemarketers faking their caller ID, I'm sure they already have some legal loophole to get through. Kinda like the Do Not Call List that doesn't work, fortunately Asterisk handles that very well. As you can tell, I'm not too keen on new laws, when the old ones don't seem to work either. On 3/29/06, Matt [EMAIL PROTECTED] wrote: Hi, Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? IE... If your name is Joe Smith you can't have Mary Smith set as the caller-id name, unless mary smith is also on your account. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to open Asterisk database
Hi, i have asterisk 1.2.5 working fine from a database in the following way: asterisk loaded as root realtime logging to a mysqld 5.x daemon listening in localhost database is astbill with username astbilluser allowed to the database the root user in mysqld has a password res_mysql.conf in /etc/asterisk has: [general] dbhost = 127.0.0.1 dbname = astbill dbuser = astbilluser dbpass = dbport = 3306 I now want to run asterisk with the -U asterisk and -G asterisk credentials. When I do it I have the error (asterisk is a valid user/group in the system with nologin as shell): /var/log/asterisk/messages: WARNING[5230] db.c: Unable to open Asterisk database WARNING[5230] db.c: Database unavailable asterisk does not exist in mysqld as a user. Asterisk loads, but cannot connect to the database, What other perms should I give it either to the mysqld or to asterisk? Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
On Wed, 29 Mar 2006 14:54:19 -0500, Matt [EMAIL PROTECTED] wrote: IE... Mary Smith seperates from Joe Smith... Joe Smith gets phone service and wants his CID to read 'Jane Smith', who is Mary's sister, so that Mary will answer when Joe calls. I was always under the impression that the telcos stored the strings associated with numbers in their own databases (or in your own cell phone), thereby only allowing you to make your name appear to be Jane Smith if you spoof your caller id to be the registered number of someone whose name is actually Jane Smith. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: AAH lost my IVR phrases
Jim Hanlon wrote: 1. The alterations to the config files made via AMP Setup pages are archived in the Asterisk DBMS, but changes made via the AMP Maintenance pages are not (Apparently. It's hard to be sure what the rules are). This is an [EMAIL PROTECTED] issue: The Setup page is provided by AMP (now called freePBX, btw), but the Maintenance page is NOT. So, the [EMAIL PROTECTED] system allows you to change configuration files built by AMP, which is where the confusion comes in. If you install Asterisk and AMP/freePBX manually, there is no maintenance tab, so there is less opportunity for you to overwrite the pre-baked configuration files. :) The rules are fairly straightforward though: Anything *_additional.conf is written by AMP/freePBX and should not be touched. Anything *_custom.conf is never touched by AMP and can be manually edited. Anything *.conf is only overwritten on upgrades of AMP, so you should take care if you edit those files. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] IAX - only one way traffic
Bjørn O wrote: In my extensions.conf I’ve got an entry for the phone number that I’m supposed to receive calls on: [default] Exten = 11223344,1,Dial(SIP/1000) exten = and not Exten = -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
Right, but SOMEONE has to have that number linked to 'Jane Smith' somewhere. I do not have any sections or numbers or dockets to quote. What I was told was that Verizon recently had regulation brought down on them that prohibits them from setting the caller-id on a number to something or someone NOT on the account. On 3/29/06, Justin Tunney [EMAIL PROTECTED] wrote: On Wed, 29 Mar 2006 14:54:19 -0500, Matt [EMAIL PROTECTED] wrote: IE... Mary Smith seperates from Joe Smith... Joe Smith gets phone service and wants his CID to read 'Jane Smith', who is Mary's sister, so that Mary will answer when Joe calls. I was always under the impression that the telcos stored the strings associated with numbers in their own databases (or in your own cell phone), thereby only allowing you to make your name appear to be Jane Smith if you spoof your caller id to be the registered number of someone whose name is actually Jane Smith. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two X100p clones. One not available for outbound?
Hi, I have an AsteriskAtHome installation with two X100p clones. Everything has been apparently fine for 5 weeks of use or so, but today, I decided to do some tweaking of my echo cancel parameters, and I realized that all along, one of my cards has been unavailable for outbound calls for some reason. Most of my calls go out one card, which is connected to a Vonage ATA. That is working fine. The other card is connected to a Verizon POTS line. This second card is working fine for inbound (I can call out from Vonage and back into VZ, and it works fine) but when I try to use my 7|. Dial pattern that ONLY lets me use the Verizon line (for testing), it says All circuits are busy. Ive tried to reboot, but its not coming back. Any hints on where to look? -Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] IAX - only one way traffic
Bjørn O wrote: This is really the day for new experiences – sorry for the load on the mailing list, but this will be the last issue I try to solve before I take the night off;) So I’ve got the incoming calls to work with a not-so-well solution (could therefore still need some feedback on the previous email). But now it appears that the IVR menu doesn’t respond to DTMF sent from my cellphone into Asterisk. I did a “iax2 debug” and there’s no particular output about dtmf keys when I press them on the cellphone, which basically tells me that the DTMF is probably passed on as audio from my provider. However, IAX would be pretty useless if there’s no way of accessing the selections in the IVR from a PSTN line. I’ve been searching the list, but have yet to find a solution to this. Thanks for all help! IAX2 does not support inband DTMF. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] IAX - only one way traffic
It's ${EXTEN} NOT {EXTEN}
Mar 29 17:44:18 NOTICE[11502]: chan_iax2.c:6794 socket_read: Rejected
connect attempt from iax.providers.server.net, who was trying to reach
'{EXTEN}@'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
___
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[Asterisk-Users] Streaming voice using IAX
Hello I have decided to use IAX to send simple voice from one end to Asterisk as IAX is more light weight than SIP. As IAX does not use RTP for media transfer, we attach the voice frames with IAX messages (miniframes may be?) libiax2 has a function called iax_send_voice that accepts following parameters: session, codec_format, data, datalen, samples Could any body point me to some documentation that explains answers to questions like: For a given codec, 1) what should be datalen 2) how often should I send data 3) what should be samples value 4) should I be concerned about byte order (say I am reading from a file and sending it over network) I am interested in sending EVRC encoded frames to Asterisk (I am not familiar with the process of sending/streaming audio frames, helpful documentation on that would also be appreciated) Thanks Jim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_odbc appears to have fields missing
On 3/29/06, Brian Roy [EMAIL PROTECTED] wrote: I'm currently using Asterisk running version 1.2.5 and trying to use cdr_odbc to connect to a Microsoft SQL database. I have everything running, but the insert statement being sent to database doesn't appear to have the start, answer, end information in it. Below is the insert statement that MS Profiler shows being sent. As you can see those fields are missing. INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES When I look at the code, in this case calldate is actually the cdr-start value. I'm working on a patch to record answer and end as well. Nathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
Nah, I don't think so, but not heard of it doesn't mean it doesnt exist, so I'll just wait and see if anyone comes up with some more info. Meanwhile I will keep spoofing as needed. :) On 3/29/06, Matt [EMAIL PROTECTED] wrote: Hi, Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? IE... If your name is Joe Smith you can't have Mary Smith set as the caller-id name, unless mary smith is also on your account. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX AAH
This question confuses me. My understanding is that FreePBX is just AMP renamed and AAH comes with AMP setup as the primary way to manage it. So, is the question realy that the user wants a newer version of AMP (read FreePBX) then the one that comes either with the newest version of AAH or the version that they have installed? Richard On 3/29/06, Dovid Bender [EMAIL PROTECTED] wrote: snip wonderful place to start. Nothing against Asterisk or Linux. My build fromscratch issues are only due to my lack of Linux experience... /snip the only way to learn is by playing. a little over a year ago i knew nothing about linux. google. is your friend. New Yahoo! Messenger with Voice. Call regular phones from your PC and save big. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FreePBX AAH
I wanted the user interface of FreePBX over what is provided in the latest version of AAH. I installed the latest version of AAH and then just installed FreePBX over the top. It went fantastic and I do like the FreePBX web interface better than the latestAAH. Thanks. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard AmermanSent: Wednesday, March 29, 2006 3:32 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] FreePBX AAH This question confuses me. My understanding is that FreePBX is just AMP renamed and AAH comes with AMP setup as the primary way to manage it. So, is the question realy that the user wants a newer version of AMP (read FreePBX) then the one that comes either with the newest version of AAH or the version that they have installed? Richard On 3/29/06, Dovid Bender [EMAIL PROTECTED] wrote: snip wonderful place to start. Nothing against Asterisk or Linux. My build fromscratch issues are only due to my lack of Linux experience... /snip the only way to learn is by playing. a little over a year ago i knew nothing about linux. google. is your friend. New Yahoo! Messenger with Voice. Call regular phones from your PC and save big. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
On 3/29/06, Matt [EMAIL PROTECTED] wrote: Hi, Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? IE... If your name is Joe Smith you can't have Mary Smith set as the caller-id name, unless mary smith is also on your account. ___ ruling by whom and in what jurisdiction? AFAIK there is no global governing body for telecom, but then I dont pay enough attention so there might be :) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with wcte11xp module
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ok when I modprobe wcte11xp I get the following message ZT_CHANCONFIG failed on channel 26: No such device or address Any ideas? Hi Jon, Did you set the E1/T1 jumper on the card correctly? for UK/Europe it should be on the E1 position. In the T1 position it would only support 25 channels (24D+1B), EuroISDN provides 31 channels (30D+1B) Other than that, check the output of lspci which should list a Network Controller like this: :00:0b.0 Network controller: Individual Computers - Jens Schoenfeld Intel 537 Does your /etc/zaptel.conf file look like this: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=uk defaultzone=uk HTH - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBRCr/D0tP/KMNOfRbAQKkDggAnVrSvkdeXuMst66ewpPti3mOaVfTaxyY L41a7uGXHjGejs/CQZBwRWDnFQ5MhDB9bHuJl2CFt/qh23pD2irMVV3tDnb6waZw HGppbuQvIvXcGyzmhzNSPgQlLLgPZnapbvRGdJ5UtKJd44IzfXSPxjBR7Mst3nyS jyEJC+eFwvcsBF9hI7mVQ9YKiDgy1namI8OCKye7pLfdOLZdb8vkM4xaUOgVlo0c 27TbOWtYyfoL8pCaPONhwvk+4vDAcE8uRShNoUV/q2Jk4LXQHaL5B3uPZjypT6rY QdL3IqDBVcizXiWVX3xXkNK4qdycYA5Q/Art2vXlpqeygjRWywr+rw== =nwqE -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with cdr_odbc
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 [EMAIL PROTECTED] wrote: My Asterisk doesn't write CDR's to database via ODBC. Plz, anybody help me to understand, what I am doing wrong. Asterisk succesfully writes CDR's into the text log. Database exists, unixODBC installed and configured properly (I can use database from isql, Asterisk reads sipusers from it...). cdr_odbc occurances in my full log: Mar 28 15:11:44 VERBOSE[8753] logger.c: [cdr_odbc.so]Mar 28 15:11:44 VERBOSE[8753] logger.c: [cdr_odbc.so] = (ODBC CDR Backend) Mar 28 15:11:44 DEBUG[8753] config.c: Parsing /usr/local/asterisk/etc/asterisk/cdr_odbc.conf Mar 28 15:11:44 DEBUG[8753] cdr_odbc.c: cdr_odbc: Logging uniqueid Mar 28 15:11:44 DEBUG[8753] cdr_odbc.c: cdr_odbc: Not logging in GMT Mar 28 15:11:44 VERBOSE[8753] logger.c: -- cdr_odbc: dsn is asterisk Mar 28 15:11:44 VERBOSE[8753] logger.c: -- cdr_odbc: username is astserver Mar 28 15:11:44 VERBOSE[8753] logger.c: -- cdr_odbc: password is [secret] Mar 28 15:11:44 VERBOSE[8753] logger.c: -- cdr_odbc: table is cdr (and that's all!) cdr_odbc.conf - [global] dsn=asterisk does this dsn match up with the entry in your odbc.ini file and does that use a driver that is in the odbcinst.ini file? - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBRCsB90tP/KMNOfRbAQJbegf/SpYCgN7RllzXsVA9wJ7IfJG7CCQ974UZ ZxGwecx+hkVw2iwbROjBht9IUBBs3w1IefRSSzqt+K4ojFKgdzOwLt43A6U8+b4n moBWUd4gD5GmWf6w8a0Z9PjDPK3+kRP2ibd8TOa/5lygZa4P+OcW8/AuVO2thDQK D2YfOQA035uvpyaDzd1D+TEI8kgQDGGLO4m8TX7t96OfM1U+CsleYmlxnBXefLsE R3G8WivYXhxgnrRkj5MiV64nPjYYSjDpEtIoIs019pG7aY4pyrLEHapLr4zHDOhr unia/S/0kB/V2wuEdIsheJT//7Vd76haYYZtqNI0lNXjcCVQyilSQw== =VJ+z -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Vonage
From what I have read, people have successfully connected directly to Vonage when they pay for the softphone option. I believe the softphone option requires a different phone number. There are setup instructions if you do a search. There is a rumor that Vonage may soon allow all accounts to work directly with IP and softphones(and therefore Asterisk) without the need to buy an ATA or softphone option number. I have not heard much about that lately. -Original Message- From: Steve Jones [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 29, 2006 5:59 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with Vonage I know Vonage doesn't officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regulatory Ruling about Caller-ID
If there was some type of ruling, its most likely to be a state public service commission (different names in different states), and not with the Fed's, etc. Right, but SOMEONE has to have that number linked to 'Jane Smith' somewhere. I do not have any sections or numbers or dockets to quote. What I was told was that Verizon recently had regulation brought down on them that prohibits them from setting the caller-id on a number to something or someone NOT on the account. On 3/29/06, Justin Tunney [EMAIL PROTECTED] wrote: On Wed, 29 Mar 2006 14:54:19 -0500, Matt [EMAIL PROTECTED] wrote: IE... Mary Smith seperates from Joe Smith... Joe Smith gets phone service and wants his CID to read 'Jane Smith', who is Mary's sister, so that Mary will answer when Joe calls. I was always under the impression that the telcos stored the strings associated with numbers in their own databases (or in your own cell phone), thereby only allowing you to make your name appear to be Jane Smith if you spoof your caller id to be the registered number of someone whose name is actually Jane Smith. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Receptionist Phones
I agree that some of these features which are considered quite basic on legacy phone systems are a major weakness on the Asterisk system. It seems to me that more time should be put into getting the basics working nicely rather than all the work going into the whiz bang bells and whistles. Just an observation not a complaint. I am not a coder so maybe I don't understand some of the challenges. -Original Message- From: John Novack [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 29, 2006 6:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Receptionist Phones The reality is, of course, that telephone systems have provided this function for many years. A DSS/BLF is available on MANY so called legacy systems, so until this function is readily available , customers that require a receptionist will continue to go elsewhere. Perhaps it is time to rethink the way data is exchanged between the CPU and the DSS/BLF? As someone said a very long time ago: Results, not excuses. JMO John Novack Christian Stredicke wrote: Well the problem with the sidecar is simple. Just try to light all lights three times within one second. If you have 50 keys there is already hell breaking loose. If you cascade side cars and say have 100 LED, this is a real Xmas tree. The CPU drowns in XML notifications. We already had trouble, and we don't want to double it at this time. Good work, IETF. BTW this is not only a problem if the phone. If the PBX has to supply 50 phones with 50 LED and e.g. they are going off hook at the same time, we are talking about a burst of 50 * 50 = 2500 messages which will have some impact of the PBX CPU as well. We need to do something about this first before we can start having 100 or 150 LED on a device. Christian - yes I am from snom. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mustardman29 Sent: Tuesday, March 28, 2006 8:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Receptionist Phones So how did the Polycom with sidecars work? I like the idea of a dedicated FOP display but not sure why you would need it if you have a Polycom with sidecars. -Original Message- From: Jerry Jones [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 28, 2006 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Receptionist Phones We installed a snom with 3 sidecars. Kinda worked, but had so many quirks they had us replace with a Polycom. All their other phones were of the poly variety. We installed a dedicated lcd running FOP for display. Receptionist was much happier. One of the key problems was she like to set the handset on her desk. But then the snom would not ring. On Mar 28, 2006, at 9:01 AM, Bob McDowell wrote: Can you chain these to get more that 42 buttons? I need about 60... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darrell Long Sent: Monday, March 27, 2006 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Receptionist Phones The 360 has an expansion unit. It adds 42 extensions. Darrell S. Long BestWeb Corporation Daniel Hazelbaker wrote: Hmm, which phone from Snom are you using for this? I've looked around their website and I can only find 3 VoIP phones, the 300, 320 and 360. The 360 by the looks of it only has 12 buttons you can assign to different extensions; am I missing something or is that the phone and you just do 12 per phone? Daniel On Mar 27, 2006, at 2:28 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yes - set up about 10 of them at a business last year. Monitoring is fine - picking up calls is a bit iffy at the best of times. (that is, picking up a ringing call by pushing the extension button. *8 works fine) Paul Hales Technical Manager AsteriskIT ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --
RE: [Asterisk-Users] Asterisk with Vonage
Linky linky: http://blog.tmcnet.com/blog/tom-keating/vonage/vonage-opens-sip-credentials. asp This is pretty cool too: (essentially free cell airtime in the continental US) http://nerdvittles.com/index.php?p=124 -Original Message- From: mustardman29 [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 29, 2006 3:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk with Vonage From what I have read, people have successfully connected directly to Vonage when they pay for the softphone option. I believe the softphone option requires a different phone number. There are setup instructions if you do a search. There is a rumor that Vonage may soon allow all accounts to work directly with IP and softphones(and therefore Asterisk) without the need to buy an ATA or softphone option number. I have not heard much about that lately. -Original Message- From: Steve Jones [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 29, 2006 5:59 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with Vonage I know Vonage doesn't officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dumb question - reaching the PSTN
Hi everyone, I am fairly new to the idea of VoIP, although I've been reading about it off and on for the last few years. Now it is starting to look mature enough to consider implementing it, but there is one thing that I haven't been able to get a clear answer on... With Vonage, you are using the Vonage network - it is their responsibility to route your call to the endpoint, which is more than likely on the old fashined PSTN. If I install Asterisk, how do my calls actually get completed? How do they get 'bridged' over to the PSTN? I attended a Seminar today hosted by Dynasis, and one of the issues was VoIP. ShoreTel was there, and the said I had to have phone lines, whether they were POTS lines, chennels from a T-1, whatever, we still had to have phone lines. Now I'm confused. If I implement an Asterisk based system (yes, I'd be paying a consultant to help), will I still have to maintain phone lines and pay full price for Long Distance? Simple pointers to White Papers on this issue will be sufficient. Many thanks, -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Between PBX and FXS
Hi guys,
I'm setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How can I figure out
in asterisk which extension was dialed before the call came to asterisk?
Does asterisk receive this information in some variable?
${BRIDGEPEER}
${CALLERID(dndi)}
${BLINDTRANSFER}
${BLINDTRANSFER}
I tried the above variables without success.
Thanks in advance.
Fernando Lujan
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Re: [Asterisk-Users] Marketing Materials
Darrick Hartman wrote: Bob McDowell wrote: The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Bob, Check on Digium's website. I know there is such a creature there. Darrick Just went looking and could not find a thing. Can you give us a url? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
