Re: [asterisk-users] Dialplan programming vs. AGI vs. ???
Yuan LIU wrote: From: Yuan LIU [EMAIL PROTECTED] But I'm curious as to the approach others use. Is doing dialplan coding in an AGI more efficient, or do people just do it that way because it's easier than learning dialplan code? Or are there some things that people think they can't do any other way? So I'm just after some ideas, really, possibly to work out if it's worth my while going down the AGI route for future projects, or not!?! Gordon, I haven't done half you have, so this is just based on what I have read (and tested) so far. You are probably asking about EAL rather than AGI. You'll need AGI only if there are functions you can't implement within Asterisk and you don't want to write a full application for Asterisk. If you are thinking about programming flexibility, EAL could be your friend because it has programming language like structures so your project remain manageable. AEL, for Asterisk Extension Language, not EAL. See ael.txt or README.ael (depending on version) in doc/ directory. Shows how little I have learned about Asterisk. Yuan Liu We have chosen to do certain funtions in AGI using PHP because we do connections to mysql and some other stuff, and and I think you have much more control with DB-related issues with AGI then with normal dialplan(.conf or AEL). However, where we dont need DB access, we only now use AEL, it really is awesome. -- thanks, Yusuf ___ --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] ELMEG IP290 and voicemail
Hello, I have Elmeg IP290 phone and have problems with VM. I don't know how to configure this IP phone, that it could call to [EMAIL PROTECTED] if I pressed VMail button. Now if I press buttom VMail , ip phone dials: sip:[EMAIL PROTECTED] (192.168.0.1 - Asterisk IP). So I don't understand , from where it takes asterisk, cause I have never write this word in configurations Maybe it takes realm... Thanks for 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] RE: Disconnected Calls
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I upgraded to the newest 1.2 Zaptel release and this is still occurring. I checked and the digium card is not sharing an IRQ with any other devices. I also changed busycount=8, and set callprogress=no. The call drops are still occurring. Mid-conversation ` in 10 calls will be disconnected. Any other suggestions? This is a relatively low volume system. Usually running less than 1 or 2 concurrent calls. Would turning on debugging logs to a file cause a problem? Many thanks, Ejay Hire Hi Ejay! Why have you excluded possibility that the problem is on telco side? -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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] Strange problem
Although I dont have an answer I would say to look at the defualt ports and see if they are opend on all sides and if NAT is used that it is set properly. - Original Message - From: Frederico Madeira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 30, 2007 2:58 PM Subject: [asterisk-users] Strange problem Hi guys. I'm working on a VOIP service provider. We have two customers running asterisk. Customer A and B. When A call to B everything is ok. When B call to A the call ring but sip messages didn't arrive on asterisk A. In my softswitch i see the invite sip message sended to A. When every other numbers(TDM and SIP) call do A everything is ok. Have any issue in asterisk that can resolve this problem ? I'm figuring out with our link provider to see if he has some firewall rules that can cause this problem I'm very very confuse becouse the invite message in every time come from my softswitch with ip of my softswitch so, why only invite originate on B side has this problem ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --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] Should I use sip gateway of PCI card?
Sangoma A200 with echo can. has been real good for me. If you need 6 FXO's I would go with one A200 with four fxo's and then a sip device for the other 2. - Original Message - From: Robert Augustyn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 30, 2007 10:26 PM Subject: [asterisk-users] Should I use sip gateway of PCI card? Hi, I am planning couple small business installations and wader what should I use for 2 to 6 lines a gateway or pci card? Any comments greatly appreciated on pros and cons and brands. Thanks, robert ___ --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] Record file name Agent
Have a look here: http://www.voip-info.org/wiki-Asterisk+variables - Original Message - From: Rafael Augusto To: asterisk-users@lists.digium.com Sent: Tuesday, January 30, 2007 8:48 PM Subject: [asterisk-users] Record file name Agent Hi people, Necessary to record agents, and that format of the archive is as below: queue-agent-exten-callerid-timestamp.wav Somebody can help me? Thanks, Rafael Rafael Augusto Gerente de Suporte Central de Relacionamento GoVoIP [EMAIL PROTECTED] www.govoip.com.br tel: mobile: 55 11 2828-2286 55 34 9997-0175 Add me to your address book... Want a signature like this? -- ___ --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 attachment: image001.jpg image002.gif Description: GIF image ___ --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 1.2.14 bristuff app_pickup.so
Hi All, I'm using Asterisk 1.2.14 with Wildcard TDM400P. I need app_pickup.so application so I can pickup channel-independent calls from any IP Phone headset. How to compile and install only this application from bristufff package? Thank you in advance, Dominik ___ --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] musiconhold restarts for every extension
On Tue, 30 Jan 2007 12:04:30 -0600 Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: While in 1.2.9 musiconhold was playing continuous on sequential extensions after a timeout, it is restarted for every extension in 1.2.14: As I understand it, this is the way Native Music on Hold works. mpg123 based MoH does not restart for each call. Well, it was working perfectly with Native MOH in 1.2.9. Judging the two replies, this is a bug(imho). I think it should at least be optional if you want it to be restarted or not(if there's anyone who needs the current behaviour). I don't want to sound like an dissatisfied customer however, i honour that asterisk is mostly voluntary work and since i'm not a programmer i try to contribute with feedback at least. Regards Thx Christian ___ --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] Regarding Call Queue
Hi I recently installed AsteriskNOW and I am trying to use its Call queue feature. But after configuring the Queue whenever I place a call, no phone in my Queue list rings. I am not able to overcome this problem. I am using Snom360 as my softphone. Please help me In this regard. With Regards Manish Gupta CAUTION - Disclaimer * This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS End of Disclaimer INFOSYS***___ --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: NAT: RTP Path Optimization
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private IP-Adresses of Int1/2
Asterisk can't give out a public IP-address for Int1/2. Where
would it get one from?
Correct that it doesn't. But some kind sould could indeed code a
variety of techniques to get it, such as:
Again: My Problem is not Intern-to-Extern (NAT,Stun). My Problem is
Extern-to-Extern, that the external phones are not talking RTP
*directly* to each other. This is bad, when Asterisk is in Europe and
the Phones are in Asia.
___
Just an idea, it's completely unverified and if I missed the point
somewhat, please excuse me ;) - but maybe this approach leads to the OPs
desired result if thought through further.
I wonder wether some clever dialplan constructs couldn't help.
I'm thinking along the lines of:
[globals]
ALLOWDIRECT=
REMAININPATH=tTwW
[internalphones] - registration context of internal phones
exten = extern1,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = extern2,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = intern1,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
exten = intern2,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
[externalphones] - registration context of internal phones
exten = intern1,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = intern2,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = extern1,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
exten = extern2,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
This assumes all sip phones are set to reinvite=yes.
I expect (one of) the options to dial (tTw or W) to force asterisk to
remain in the media path. This way *only* if it's int-int or ext-ext
will it send sip reinvite, right?
I have another idea just now, but that's even weirder ... ;)
Conrad
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[asterisk-users] Hi Honies! I'm home!
Many of you may have seen the recent announcement about Danny Windham coming on as the new CEO of Digium. This is one of the most exciting things to happen to Digium and to Asterisk at large. When Danny comes on board, I will be transitioning to the role of Chief Technical Officer (retaining my position of chairman of the board of directors), providing strategic vision for the company as well as being able to focus more extensively on the community, the customers and the technology. My sincere hope is that this transition will not only directly benefit the Asterisk community and Digium customers, but will allow me to spend much more time with the community and with Asterisk, playing a more important technical role in our roadmap for both hardware and software. I'm looking forward to working more with the community and the developers to help grow the future of Asterisk even more! Mark ___ --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: NAT: RTP Path Optimization
On Wednesday 31 January 2007 8:28 am, Conrad Wood wrote: This assumes all sip phones are set to reinvite=yes. I expect (one of) the options to dial (tTw or W) to force asterisk to remain in the media path. This way *only* if it's int-int or ext-ext will it send sip reinvite, right? Yes, but now you have to be careful of unintended consequences when people are trying to use IVRs. What's wrong with having two peers, one with canreinvite=no, and Dial() using the appropriate one? (I haven't been following the thread, so this may have already been discussed, and discounted.) -A. ___ --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: NAT: RTP Path Optimization
On Wed, 2007-01-31 at 08:42 -0500, Andrew Kohlsmith wrote: On Wednesday 31 January 2007 8:28 am, Conrad Wood wrote: This assumes all sip phones are set to reinvite=yes. I expect (one of) the options to dial (tTw or W) to force asterisk to remain in the media path. This way *only* if it's int-int or ext-ext will it send sip reinvite, right? Yes, but now you have to be careful of unintended consequences when people are trying to use IVRs. I wouldn't take it live as is without further testing, but I guess the idea was worth adding to the thread. What's wrong with having two peers, one with canreinvite=no, and Dial() using the appropriate one? (I haven't been following the thread, so this may have already been discussed, and discounted.) That was the other idea I had - but think different Dial parameters are less problematic. For 2 peers per phone, the phones either need to have a static public IP or need to be able to register with 2 credentials (e.g. snom/cisco). Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Dial Plan
Perfect. Here's a quick and hopefully doable followup question. We have Polycom Soundpoint 501 phones. Is there a way to have a phone check 2 voicemail boxes? If we have a queue, and we want the MWI to show for say that users's extension 1000 and the special billing vm box of 2000. Either way, they'll all get the email (its a group email), but it would be nice to have the light as well. Rob Lee Jenkins wrote: Rob Schall wrote: A question about Queues and Dial Plans We are trying to set up a customer service queue. I've set up the queue and agents who will participate. However, there's still one area I'm not sure how to make it work. After 60 seconds, I need it to decide that no one is available, and forward it to an email box of my choosing. Is this possible? Rob Have you tried setting the timeout parameter? http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue exten=1,1,Queue(support,t|||60) exten=1,2,Voicemail(123124125) ___ --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] Regarding Call Queue
Is your agent logged into that queue to receive the calls? You can typically say show queues to list all queues and see who is not in use vs unavailable. If they are all unavailable, are you getting a successful Agent Logged In message when you log that guy in? Rob Manish Gupta02 wrote: Hi I recently installed AsteriskNOW and I am trying to use its Call queue feature. But after configuring the Queue whenever I place a call, no phone in my Queue list rings. I am not able to overcome this problem. I am using Snom360 as my softphone. Please help me In this regard. With Regards Manish Gupta CAUTION - Disclaimer * This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS End of Disclaimer INFOSYS*** ___ --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] pickup internal and external calls
hello, i want to make a dialplan where i can pickup calls to an extension when there are internal and external calls. i want to use only one prefix for pickup both situations so there is a plan how to check if the incoming call is an internal call or an extern??? regards rene ___ --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] pickup internal and external calls
Please define pickup. Do you want to get parked calls are or you looking to send all calls to a specific phone ? - Original Message - From: René Enskat To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, January 31, 2007 4:39 PM Subject: [asterisk-users] pickup internal and external calls hello, i want to make a dialplan where i can pickup calls to an extension when there are internal and external calls. i want to use only one prefix for pickup both situations so there is a plan how to check if the incoming call is an internal call or an extern??? regards rene -- ___ --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] Hi Honies! I'm home!
I was wondering when this would happen. A lot of successful and prospering open source company like yours seems to do this. Much like Google did. Once a company has grown to a point it's more valuable to have someone focus on the business from a businessmans perspective working with the monies, departments, board of directors and strategies while letting the previous guru (Mark) focus on what you always really have needed to, the code and the product line. It looks like Danny has a solid background and strong roles of leadership from adtran. I love this decision. Go team Digium. Brian On 1/30/07, Mark Spencer [EMAIL PROTECTED] wrote: Many of you may have seen the recent announcement about Danny Windham coming on as the new CEO of Digium. This is one of the most exciting things to happen to Digium and to Asterisk at large. When Danny comes on board, I will be transitioning to the role of Chief Technical Officer (retaining my position of chairman of the board of directors), providing strategic vision for the company as well as being able to focus more extensively on the community, the customers and the technology. My sincere hope is that this transition will not only directly benefit the Asterisk community and Digium customers, but will allow me to spend much more time with the community and with Asterisk, playing a more important technical role in our roadmap for both hardware and software. I'm looking forward to working more with the community and the developers to help grow the future of Asterisk even more! Mark ___ --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 -- --Brian McManus --(208) 329-0818 ___ --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] Testing IVR / Callcenter applications
Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards ___ --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] Testing IVR / Callcenter applications
Why don´t you put the IVR in an extension... and call it also from an extension of the same PBX. On 1/31/07, fadi mujahid [EMAIL PROTECTED] wrote: Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards ___ --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] Testing IVR / Callcenter applications
Thanks for ur suggestion. But the problem is that won't test the queuing of the outbound and inbound calls of the callcenter thanks again On 1/31/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Why don´t you put the IVR in an extension... and call it also from an extension of the same PBX. On 1/31/07, fadi mujahid [EMAIL PROTECTED] wrote: Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards ___ --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] Asterisk dual contexts stupidity
Significant albeit insanely stupid Asstricks message:
2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got
something to jump out with ('2')!
(Oooh how about creating errors we can figure out Digium!)
Any thoughts
What Error ? it says DEBUG
This just tell you that the user pressed '2'
Actually, the first time I read that message I was laughing :)
Only in opensource product you have the priviledge of having funny message
hth
___
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Re: [asterisk-users] Dialplan programming vs. AGI vs. ???
Gordon Henderson wrote: Just a general question on dialplan programming... I've implemented a fairly full-featured system using dialplan code only. I've not used any AGI for it, yet it ticks all the boxes I want it to tick (diverts, follow-me, voicemail, dnd, outdialing restrictions, simple auto-attendant, and numerous star codes to control it all) This is all aimed at the small/medium office PBX type application. But I'm curious as to the approach others use. Is doing dialplan coding in an AGI more efficient, or do people just do it that way because it's easier than learning dialplan code? Or are there some things that people think they can't do any other way? So I'm just after some ideas, really, possibly to work out if it's worth my while going down the AGI route for future projects, or not!?! Any feedback is most welcome! Cheers, Gordon I've only been using Asterisk for a short while, but have been programming for about 10 years so AEL appeals to me. Steve Murphy has done an outstanding job on AEL2. But IMO it all depends on the job at hand. For instance, I wanted to be able to access FirebirdSQL databases from the dialplan and the only viable way was through AGI. My personal thought (and practice) has been: 1. If it's dialplan specific (Dial(),Playback(), etc) then Asterisk script, preferably AEL2. 2. Even if it's dialplan specific, but prone to require any appreciable resources, off load it to an AGI. 3. If it's not dialplan specific (FirebirdSQL access, SOAP calls, etc) then definitely off load it to AGI. Remember there is also FastAGI which allows us to scale a system by off loading resource intensive stuff to other computers entirely when the situation requires it. Personally, I'm glad that there is so many different ways to interact with Asterisk. Nice having a swiss army knife ;) -- Warm Regards, Lee ___ --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] Testing IVR / Callcenter applications
What do you mean? Setup another box, make a bunch of calls (as if you were clients) into the production box, use back to back E1 cards. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of fadi mujahid Sent: Wednesday, January 31, 2007 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Testing IVR / Callcenter applications Thanks for ur suggestion. But the problem is that won't test the queuing of the outbound and inbound calls of the callcenter thanks again On 1/31/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Why don´t you put the IVR in an extension... and call it also from an extension of the same PBX. On 1/31/07, fadi mujahid [EMAIL PROTECTED] wrote: Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards ___ --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] Testing IVR / Callcenter applications
We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? Just use an IAX or SIP thrunk to/from another Asterisk. there is no real difference from Asterisk's stand point if the call comes from IAX, SIP or ZAP hth ___ --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] Line drops strange problem(got event On hook)
Hello to all, I have a strange problem with my asterisk. Line drops while i am in a call and without a reason.The line drops no matter if it is a incoming or outgoing call and it happen while i am talking on the phone (no silence detection problem). I tried to debug the situation and the only strange thing is the got event On hook (i guess..). I am thinking that it is a problem with the card TDM400P or a hardware problem. Anyone who can help?? thank you in advance p.s. i put the debug messages in case someone wants to take a look. It is while i got a call and after a while the line dropped. Jan 31 15:20:40 VERBOSE[25962] logger.c: -- SIP/51-0986fab0 is ringing Jan 31 15:20:40 DEBUG[25962] chan_zap.c: Requested indication 3 on channel Zap/7-1 Jan 31 15:20:40 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:20:40 DEBUG[25962] chan_zap.c: Got event Ring Begin(18) on channel 7 (index 0) Jan 31 15:20:42 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:20:42 DEBUG[25962] chan_zap.c: Got event Ring/Answered(2) on channel 7 (index 0) Jan 31 15:20:42 DEBUG[25962] chan_zap.c: Setting IDLE polarity due to ring. Old polarity was 0 Jan 31 15:20:45 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:20:45 DEBUG[25962] chan_zap.c: Got event Ring Begin(18) on channel 7 (index 0) Jan 31 15:20:46 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:20:46 DEBUG[25962] chan_zap.c: Got event Ring/Answered(2) on channel 7 (index 0) Jan 31 15:20:46 DEBUG[25962] chan_zap.c: Setting IDLE polarity due to ring. Old polarity was 0 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Setting NAT on RTP to 524288 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 52547: Match Found Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Setting NAT on RTP to 524288 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Checking SIP call limits for device 53 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED] Jan 31 15:20:47 DEBUG[2434] channel.c: Avoiding initial deadlock for 'SIP/53-b7a05818' Jan 31 15:20:47 DEBUG[2442] channel.c: Planning to masquerade channel SIP/53-b7a05818 into the structure of SIP/51-0986fab0 Jan 31 15:20:47 DEBUG[2442] channel.c: Done planning to masquerade channel SIP/53-b7a05818 into the structure of SIP/51-0986fab0 Jan 31 15:20:47 DEBUG[25962] channel.c: Got clone lock for masquerade on 'SIP/53-b7a05818' at 0xb7a0adf4 Jan 31 15:20:47 DEBUG[25962] chan_sip.c: update_call_counter(51) - decrement call limit counter Jan 31 15:20:47 DEBUG[25962] chan_sip.c: Acked pending invite 102 Jan 31 15:20:47 DEBUG[25962] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jan 31 15:20:47 DEBUG[25962] channel.c: Putting channel SIP/53-b7a05818 in 64/64 formats Jan 31 15:20:47 DEBUG[25962] channel.c: Released clone lock on 'SIP/51-0986fab0ZOMBIE' Jan 31 15:20:47 DEBUG[25962] channel.c: Done Masquerading SIP/53-b7a05818 (0) Jan 31 15:20:47 VERBOSE[25962] logger.c: -- SIP/53-b7a05818 answered Zap/7-1 Jan 31 15:20:47 DEBUG[25962] chan_zap.c: Requested indication -1 on channel Zap/7-1 Jan 31 15:20:47 DEBUG[25962] chan_zap.c: Took Zap/7-1 off hook Jan 31 15:20:47 DEBUG[25962] chan_zap.c: Enabled echo cancellation on channel 7 Jan 31 15:20:47 DEBUG[25962] chan_zap.c: Engaged echo training on channel 7 Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Not Found Jan 31 15:20:47 DEBUG[2442] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 52548: Match Found Jan 31 15:22:40 DEBUG[25962] chan_zap.c: Exception on 19, channel 7 Jan 31 15:22:40 DEBUG[25962] chan_zap.c: Got event On hook(1) on channel 7 (index 0) Jan 31 15:22:40 DEBUG[25962] chan_zap.c: disabled echo cancellation on channel 7 Jan 31 15:22:40 DEBUG[25962] channel.c: Didn't get a frame from channel: Zap/7-1 Jan 31 15:22:40 DEBUG[25962] channel.c: Bridge stops bridging channels Zap/7-1 and SIP/53-b7a05818 Jan 31 15:22:40 DEBUG[25962] chan_sip.c: update_call_counter(53) - decrement call limit counter Jan 31 15:22:40 DEBUG[25962] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jan 31 15:22:40 VERBOSE[25962] logger.c: == Spawn extension (ringoffice, s, 1) exited non-zero on 'Zap/7-1' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is '00381113237515' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is '00381113237515' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is 's' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is 'ringoffice' Jan 31 15:22:40 DEBUG[25962] pbx.c: Function result is 'Zap/7-1' Jan 31 15:22:40 DEBUG[25962] pbx.c:
Re: [asterisk-users] Dialplan programming vs. AGI vs. ???
Lee Jenkins wrote: Gordon Henderson wrote: Just a general question on dialplan programming... I've implemented a fairly full-featured system using dialplan code only. I've not used any AGI for it, yet it ticks all the boxes I want it to tick (diverts, follow-me, voicemail, dnd, outdialing restrictions, simple auto-attendant, and numerous star codes to control it all) This is all aimed at the small/medium office PBX type application. But I'm curious as to the approach others use. Is doing dialplan coding in an AGI more efficient, or do people just do it that way because it's easier than learning dialplan code? Or are there some things that people think they can't do any other way? So I'm just after some ideas, really, possibly to work out if it's worth my while going down the AGI route for future projects, or not!?! Any feedback is most welcome! Cheers, Gordon I've only been using Asterisk for a short while, but have been programming for about 10 years so AEL appeals to me. Steve Murphy has done an outstanding job on AEL2. But IMO it all depends on the job at hand. For instance, I wanted to be able to access FirebirdSQL databases from the dialplan and the only viable way was through AGI. My personal thought (and practice) has been: 1. If it's dialplan specific (Dial(),Playback(), etc) then Asterisk script, preferably AEL2. 2. Even if it's dialplan specific, but prone to require any appreciable resources, off load it to an AGI. 3. If it's not dialplan specific (FirebirdSQL access, SOAP calls, etc) then definitely off load it to AGI. Remember there is also FastAGI which allows us to scale a system by off loading resource intensive stuff to other computers entirely when the situation requires it. Personally, I'm glad that there is so many different ways to interact with Asterisk. Nice having a swiss army knife ;) Could'nt have said it better! -- thanks, Yusuf ___ --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: Enterprise quality SIP provider
You can try us, http://www.voicemeup.com TDM in most areas , others offloaded white routes to L3 mainly. Cover most of usa , and canada. you can ping www.voicemeup.com to get an idea on location , we are directly on peer1,teleglobe,videotron with best quality bandwith only. Per minute pricing starts at 0.019 and goes down to 0.009 on volume, automatic and realtime adjustments starting at 2500 minutes. On 1/30/07, Martin Joseph [EMAIL PROTECTED] wrote: On 2007-01-28 08:37:43 -0800, Eric Germann [EMAIL PROTECTED] said: We LOVE Teliax. We're on a Time Warner business class fiber connection and avg 25ms latency from Ohio to Denver CO. With that connection I would love Teliax also. Marty ___ --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 -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --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] Testing IVR / Callcenter applications
You can use a cross-over cable between Asterisk boxes to imitate the functionality of a T/E-1. Bill Gibbs wrote: What do you mean? Setup another box, make a bunch of calls (as if you were clients) into the production box, use back to back E1 cards. Bill *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *fadi mujahid *Sent:* Wednesday, January 31, 2007 10:34 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Testing IVR / Callcenter applications Thanks for ur suggestion. But the problem is that won't test the queuing of the outbound and inbound calls of the callcenter thanks again On 1/31/07, *Alejandro Lengua* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Why don´t you put the IVR in an extension... and call it also from an extension of the same PBX. On 1/31/07, fadi mujahid [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 Voipjet ...
That's interesting I use Voipjet cheap lines and I don't have a problem at all. Peter On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Hello, we have this problem with Trixbox 1.23 I have created an outgoing route where the 1st line has Voipjet and the 2nd an 3rd have voipcheap accounts. The problem is that at certain moments, when we call all the calls go through the voipcheap SIP accounts SIP, whose quality are not only not good enough but also consume a lot of bandwidth. The error message that returns Voipjet to Asterisk is that all circuits busy. What I asume from this? Thanks in advance ___ --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] Dynamically Adding A Context
Seriously man. I don't want to be testy here, but what part of *dynamic* didn't you understand? Adding a context to a flat file and reloading the server is NOT dynamic. And, as I explained in a previous post, realtime is not a solution I can use for this issue because I'm updating proxy software that uses the AMI so realtime is not an option. For everyone else; Thanks for trying to take a stab at this. It seems there simply is no way to do it. Perhaps I'll submit a patch to digium so at least we have this simple functionality in the future... j On Tue, 2007-01-30 at 13:17 -0900, Shane Spencer wrote: Reload.. Reload.. Reload.. On 1/30/07, Shane Spencer [EMAIL PROTECTED] wrote: Realtime.. Realtime.. Realtime.. On 1/30/07, j [EMAIL PROTECTED] wrote: On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote: On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote: In order to do this, I have to add a couple quick extensions to the dial plan dynamically, so I have to be able to add my own context. from API use Command to run the CLI command add extension But you can only add to an existing context with that. Yes exactly. I tried the 'add extension' command. With *and* without the 'replace' argument, if the context does not already exist the command gives an error ;( ___ --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] (no subject)
hi every body, i m new to this mail list, and also with asterisk IPBX, i havr digium TE110P card, can someone till me if he has an particular experience with this card, kind of bugs, problems... kind regards Younss ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Dial Plan
Rob Schall wrote: Perfect. Here's a quick and hopefully doable followup question. We have Polycom Soundpoint 501 phones. Is there a way to have a phone check 2 voicemail boxes? If we have a queue, and we want the MWI to show for say that users's extension 1000 and the special billing vm box of 2000. Either way, they'll all get the email (its a group email), but it would be nice to have the light as well. I'm not sure about that one as I have not had the need for it yet. I know that you can send the same voicemail to several boxes at the same time. Like the example that I gave before. exten=s,1,Voicemail(123124125) ... would send the voicemail to boxes 123, 124 and 125. -- Warm Regards, Lee ___ --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] Testing IVR / Callcenter applications
Hello, We usually use a crossover T1/E1 cable and a multi-port T1/E1 card and call the server from itself or another Asterisk server. We have used this method to do stress testing in VICIDIAL, which has a builtin set of tools for stress testing outbound dialing. MATT--- On 1/31/07, fadi mujahid [EMAIL PROTECTED] wrote: Thanks for ur suggestion. But the problem is that won't test the queuing of the outbound and inbound calls of the callcenter thanks again On 1/31/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Why don´t you put the IVR in an extension... and call it also from an extension of the same PBX. On 1/31/07, fadi mujahid [EMAIL PROTECTED] wrote: Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards ___ --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] Testing IVR / Callcenter applications
you can use a SIP based phone service to try it out Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! MSN: [EMAIL PROTECTED] Skype: Rehan33 First they ignore you, then they laugh at you, then they fight you, then you win. By Mahatma Gandhi. ___ --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] Testing IVR / Callcenter applications
Hi, We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? I just begun to think how to do the same thing... but considering a Cisco infrastructure (CallManager, IPIVR, voice gateway/router, proggers...) Is there any way i can trigger a bunch of calls to the cisco callmanager (and then to the IVR). Ideally, i was thinking about something scripteable, so i can extract processing times and so. Sorry if i sound like a newbie... it is because i am a newbie (first time on voip, and just discovered open source voip packages). Thanks very much in advance! -- Saludos, miguel Los agujeros negros son lugares donde dios dividió por cero. Black holes are places where god divided by zero. ___ --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] Toll-free dialing via PRI problem
Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but the calls are never answered. All other calls, and most toll-free numbers are not affected. The numbers that are affected are all travel related companies (United Airlines, American Airlines, US Air, Starwood Hotels, etc.) we cannot connect to any of these numbers. Hey Tim, All I can offer you is the fact that I see the exact same thing on my setup that uses * and a TE411P. I've also seen it when calling Lenovo tech support and Sirius Satellite Radio. On the latter two, it bypasses the auto-attendant when I call and connects me straight to an operator/technician. When you call on regular PBX or cell phone, you are greeted by an auto-attendant, press 1, yada-yada. Let us know what you find out. Cheers, Stefano ___ --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] Storing recordings
Hello, I'm currently facing a decision regards to the system I have to build. Basically, I'm aiming for 2 Asterisk servers with 1 PRI line in each. And each of them will record all calls in and out. I was wondering if anyone had any suggestions in that regards ? I'm currently thinking of building these 2 servers, with some Dell PowerEdge 2950 and either connected directly to a Gigabit switch that would be connected to another machine that will have a lot of drive space. I could maybe mount some network drives on the 2950. Or maybe I could connect the 2950 to the other machine with a Fiber cable ? But i'm still undecided if I should use a SAN or NAS. I guess that in overall and if I wanna build something though enough to stay there for a least 3-4 years, I should go for a SAN. Anyone any thoughts ? First time I'm building a such system, I have good knowledge about it since I already used some in the past, but if anyone has experience with Asterisk and such system it would be great to share your experience! Thanks! ___ --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] Queue Status
Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk? If this call is coming from a queue, do not follow a normal dial plan for the phone (don't send to user's voicemail). In stead, once timed out (t|||60), send to Voicemail(u1000). Lee recommended QUEUESTATUS, but that seems to return if anyone is in a specific queue, and not if the current call came from a queue. I probably just misunderstand how it all works. :) Thanks all! Rob - I would recommend that you download the following tool and play with it (if you have a windows box): http://www.datatrakpos.com/pos/datatalk/Default.aspx Check out the Visual Menu Builder included. There is a Queue widget included. Try playing around with that and building the project and inspecting the resulting script that the program generates. This should give you a better idea of how to do what you are trying to do, I think. Also, check out the channel variable QUEUESTATUS. See if it's set to determine whether the call is coming from a queue. I'm not sure if it's normally blank or has a default value so you may want to check that out too. ___ --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] musiconhold restarts for every extension
Lacy Moore - Aspendora wrote: On 1/30/07, *Benko* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello! I've upgraded from 1.2.9 to 1.2.14 recently but experience an unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was playing continuous on sequential extensions after a timeout, it is restarted for every extension in 1.2.14: The powers that be decided this was better. Personally, I'd rather it not restart, but, like most things in life, that is not my decision to make, nor do I have the talent to change this behavior. I take it, then that the correct solution to this problem is to use mpg123 for music-on-hold? This really should be configurable. -Stephen- ___ --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] Compiling NVFaxDetect and other Newman apps on Asterisk 1.4
If you are having problems compiling NVFaxDetect (app_nv_faxdetect.c) or other Newman Telecom applications on Asterisk 1.4, please look at Steve's comments at: http://www.voip-info.org/wiki/view/NewmanTelOnAsterisk14 Several changes to Asterisk prevents NVFaxDetect and other apps from registering. Some changes needed. He also has copies of the code if you need it... Justin Newman Food fight? Enjoy some healthy debate in the Yahoo! Answers Food Drink QA. http://answers.yahoo.com/dir/?link=listsid=396545367 ___ --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: [asterisk-dev] Dynamically Adding A Context
yusuf wrote: j wrote: Greetings! I've searched far and wide for an answer and have gotten no where, so I was hoping one of you guys might have the answer; Is it possible to dynamically add a context to the dialplan? You can add extensions via the CLI, however if the context doesn't exist I get an error message instead of it creating the context for me. Any method will do, AGI, AMI, CLI... I just need a solution :) Yusuf, I was just curious what kind of context do you need to add? The reason I ask is maybe you could use a custom AGI script to simulate the same steps that would occur in a dynamically added context? You could for instance, create an AGI that reads from a dynamically created text file template or flat script? Just a thought... -- Warm Regards, Lee ___ --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 Voipjet ...
How many simultaneous calls per account are you sending ? On 1/31/07, Peter Halliday [EMAIL PROTECTED] wrote: That's interesting I use Voipjet cheap lines and I don't have a problem at all. Peter On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Hello, we have this problem with Trixbox 1.23 I have created an outgoing route where the 1st line has Voipjet and the 2nd an 3rd have voipcheap accounts. The problem is that at certain moments, when we call all the calls go through the voipcheap SIP accounts SIP, whose quality are not only not good enough but also consume a lot of bandwidth. The error message that returns Voipjet to Asterisk is that all circuits busy. What I asume from this? Thanks in advance ___ ___ --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] Dynamically Adding A Context
At 02:17 PM 1/30/2007, you wrote: Yes exactly. I tried the 'add extension' command. With *and* without the 'replace' argument, if the context does not already exist the command gives an error ;( You could create a set of empty extensions to use and re-use as needed. It's one of those tasks that made you wish array() had something to do with arrays!!! Ira ___ --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] E911 Bill Announced
Nelson, Clinton, Snowe REIntroduce Voice Over Internet E-911 Legislation Bill Will Prevent Tragedies By Making Sure Calls for Help Made On Internet-Based Telephone Service Connect to Local 911 Washington, DC - Senator Bill Nelson (D-FL); Senator Hillary Rodham Clinton (D-NY), Co-Chair of the Congressional E-911 Caucus and Senator Olympia Snowe (R-ME) today announced that they have reintroduced the IP Enabled Voice Communications and Public Safety Act. The bill addresses the need to ensure the growing number of Voice Over Internet Protocol (VoIP) telephone service subscribers have full access to 911, including Enhanced (E)-911 capability that allows 911 dispatchers to trace the phone number and location of calls for help. Unfortunately, we ve seen the tragic consequences when consumers can t connect to 911 services through their Internet phone company, Senator Nelson said. VoIP subscribers should feel confident that they will have access to emergency services it could be a matter of life or death. It is critical that the millions of households using this technology can reach 911 when tragedy strikes. All emergency calls, whether made on a land line, cell phone or Internet-based phone service, need a rapid response. It could truly make the difference in saving a life, said Senator Clinton. The inability of the emergency response network to keep pace with voice over Internet protocol technology has left millions of VoIP subscribers without guaranteed access to emergency services, Senator Snowe said. Innovation and technological advances should improve the lives of Americans, not endanger them. VoIP subscribers should not be susceptible to substandard emergency service simply because they are on the cutting edge of in home telecommunications technology. VoIP telephone customers are connected to broadband internet lines instead of traditional phone lines. Ensuring that 911 calls made from VoIP phones are properly routed and responded to has presented new challenges to public safety officials. There have been several tragedies in which VoIP 911 calls were either routed to closed business offices instead of emergency dispatcher or could not be connected. The Clinton-Snowe-Nelson bill will allow VoIP companies to patch into the 911 networks operated by the traditional phone companies. The bill also ensures that consumers are fully informed if their VoIP provider cannot ensure that their 911 call will be properly routed in an emergency. Furthermore, the legislation tasks the National E-911 Implementation Coordination Office -- created under the ENHANCE Act introduced by Senator Clinton and signed into law in 2004 -- to develop a plan for a nationwide network and make recommendations to Congress in order to ensure that all 911 VoIP calls are responded to properly. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Status
In the queues that I've established, I've assigned a different number to queue-agents than their normal extension. If their extension is 2120, their roll-over (second extension) would be 3120 and their queue-agent-id would be 4120. That way I can assign a different dial-plan for 4120 that doesn't include voicemail (or if it does, it's the queue's voicemail rather than the individual agent's voicemail). Hope that helps. Rob Schall wrote: Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk? If this call is coming from a queue, do not follow a normal dial plan for the phone (don't send to user's voicemail). In stead, once timed out (t|||60), send to Voicemail(u1000). Lee recommended QUEUESTATUS, but that seems to return if anyone is in a specific queue, and not if the current call came from a queue. I probably just misunderstand how it all works. :) Thanks all! Rob - I would recommend that you download the following tool and play with it (if you have a windows box): http://www.datatrakpos.com/pos/datatalk/Default.aspx Check out the Visual Menu Builder included. There is a Queue widget included. Try playing around with that and building the project and inspecting the resulting script that the program generates. This should give you a better idea of how to do what you are trying to do, I think. Also, check out the channel variable QUEUESTATUS. See if it's set to determine whether the call is coming from a queue. I'm not sure if it's normally blank or has a default value so you may want to check that out too. ___ --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] Toll-free dialing via PRI problem
This is a common issue with large inbound call center operations. They like to cheat. They actually start sending prompts to the caller without actually signalling their carrier that they have answered the line. Typically they do not answer until a phone is ringing or you are in a queue. I do believe this is illegal per the FCC. From asterisk, you do not hear anything other than ringing as it does not cut the audio path through until it receives the answer from the far end, hence the steady ringing. This allows the large centers to reduce their billable minutes by enough to warrent them to try it. On Jan 31, 2007, at 10:51 AM, McGhee, Stefano wrote: Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but the calls are never answered. All other calls, and most toll-free numbers are not affected. The numbers that are affected are all travel related companies (United Airlines, American Airlines, US Air, Starwood Hotels, etc.) we cannot connect to any of these numbers. Hey Tim, All I can offer you is the fact that I see the exact same thing on my setup that uses * and a TE411P. I've also seen it when calling Lenovo tech support and Sirius Satellite Radio. On the latter two, it bypasses the auto-attendant when I call and connects me straight to an operator/technician. When you call on regular PBX or cell phone, you are greeted by an auto-attendant, press 1, yada-yada. Let us know what you find out. Cheers, Stefano ___ --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] Help with semaphores
I'm looking for some help from any Asterisk heavy who might be doing
something similar to what I'm trying to do...
Background:
I work for a research lab, testing telephony products and tools.
Historically, we used Ameritec Crescendos and Fortissimos to act as load
generators and call sinks when testing equipment. However, the
equipment we are testing gets more and more complex, and the scripted
scenarios the Ameritecs give have become a limiting factor for testing.
Therefore, Asterisk was chosen as a possible solution (we're a cheap lab).
I've been learning Asterisk as I go, but I've learned a lot. Here's the
basic scenario:
We are using an Asterisk (AAH 2.8, specifically) to sink calls. I do
this by taking the ${EXTEN} and breaking it down by sections until I get
to the last 4 digits (i.e., 2105551212). Once I get to the 4-digit
extension, I am trying to set a flag, or semaphore, to do Busy/Idle
testing. Here is my extensions_custom.conf fragment:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at the
beginning of [from-trunk-custom] to the full dialed digits in ${EXTEN},
before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has already
called this number, return busy
exten = 1212,102,System(echo ${UNIQUEID} /tmp/${orig_num}) ;
basically, create a file in /tmp whose name is the full number from the
beginning. In this case, the full
; filename would be /tmp/2105551212. I don't really care about the
contents, though.
exten = 1212,103, Goto(Idle,1) ; from here, we jump to a new extension
called Idle, where we do a Random to decide whether to simulate no one
home (ring no answer) or
; we send ring for
about 10 seconds, then Answer() and play some .wav files, then hangup.
The last thing we do in either case is to delete
; the
/tmp/${orig_num} file.
The above code works very well at low call volumes. However, I'm
running into race conditions at high call volumes where several calls
are getting through the test in priority 1 before the file is created in
priority 102 (n+101).
I've tried to implement semaphores by using both local and global
variables, but it doesn't seem to work.
My ultimate question: Is anyone doing something similar, and what did
you do to implement the busy/idle.
I appreciate any help anyone can offer.
Mitch Thompson
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Re: [asterisk-users] Timeout in IAX vs SIP
30 jan 2007 kl. 06.38 skrev Yuan LIU: When Asterisk dials an IAX destination with no registration, it very quickly comes to the conclusion that it can't make the call -- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, IAX2/ [EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED]/[EMAIL PROTECTED] [Jan 29 21:43:15] NOTICE[1957]: chan_iax2.c:2686 __auto_congest: Auto-congesting call due to slow response -- IAX2/216.207.245.8:4569-1 is circuit-busy -- Hungup 'IAX2/216.207.245.8:4569-1' == Everyone is busy/congested at this time (1:0/1/0) But if Asterisk Dials a SIP destination it doesn't have a registration, it waits for a very long time before giving up. What is the difference? Does IAX use TCP instead of UDP? Is there some way to change timeout value in SIP attempt so it gives up in a reasonable time? Both protocols use UDP, bot the timers are a bit different. However, if there's no registration both channels should act the same unless there's a configuration that's giving wrong information to chan_sip, like you having a username= or defaultip= setting. /O ___ --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: Multiple parking lot
Check what's going on in that branch. I also believe there's an open issue in the bug tracker for this, so you can see when it's ready for testing again. Thanks, /Olle 31 jan 2007 kl. 07.04 skrev Tim Ferguson: Is there any chance you could contact me or give me a website to monitor the current status of implementing multiple parking lots. Multiple parking lots in 1.4 is something we were hoping for and would love to see happen. I'd be happy to help with testing/debuging. Please feel free to contact me. On Fri, 26 Jan 2007 08:28:50 +0100, Olle E Johansson [EMAIL PROTECTED] wrote: 25 jan 2007 kl. 08.26 skrev Darryl Dunkin: There is an SVN branch with this feature: http://svn.digium.com/view/asterisk/team/oej/multiparking/ I had hope this would be a feature added to Asterisk 1.4, but fail to see it on the changelog. It wasn't approved due to some architecture issues. I'll see if I get time to fix them for next release. /O ___ --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 --- * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ --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 dual contexts stupidity
Time Bandit wrote:
Significant albeit insanely stupid Asstricks message:
2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got
something to jump out with ('2')!
(Oooh how about creating errors we can figure out Digium!)
Any thoughts
What Error ? it says DEBUG
This just tell you that the user pressed '2'
Actually, the first time I read that message I was laughing :)
Only in opensource product you have the priviledge of having funny message
hth
I was once working on tracking down a particularly elusive bug in one of
our products and put a small piece of code showing a message when
testing a value that said Sh*t still doesn't work if a certain value
was true.
I guess you already know that I forgot to remove this little tidbit
before distributing the update.
Needless to say, another update was fast following...
--
Warm Regards,
Lee
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Re: [asterisk-users] Re: NAT: RTP Path Optimization
Thanks for this discussion! I've gotten a few ideas for better NAT handling in chan_sip3. The current way is implemented in so many installations, so it would be hard to turn it around, but in pineapple I can freely break backwards compatibility. Let me think about it for a few days, then I'll try to summarize. What's Pineapple and chan_sip3? Check http://www.codename-pineapple.org Major update coming soon, hopefully. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Status
Hi Rob,
put your call centre stuff in a context that is separate from all other
extensions (like internal, long distance, etc) and have it contain it's
own, dedicated dial() code.
[incoming-to-callcentre]
; Incoming calls to Call Centre arrive in this context
; IVR stuff.
;
; If Q1 selected...
exten = 1,1,Goto(5210,1)
; If Q2 selected...
exten = 2,1,Goto(5220,1)
include = [callcentre]
[call-centre]
; All Call Centre dial stuff goes here
; Q1
exten = 5210,n,Queue(Q1)
; Q2
exten = 5220,n,Queue(Q2)
; All CC extensions start with 6
exten = _6XX,1,macro(ccexten,${EXTEN})
[macro-ccexten]
; Special dial stuff for Call Centre only
exten = s,1,Dial(EXTEN)
; Handle timeouts, etc here ...
exten = s,n,Voicemail(u1000)
regards,
Drew
Rob Schall wrote:
Hello all,
I think Lee has given me a head start, but I'm still running in a circle
(at least i'm in the lead).
The problem is with my queues. The phones go to their own voicemail
after 5 rings.
That's about the same time I allow the phone to ring before trying
another phone in the queue. Is there a way to tell asterisk?
If this call is coming from a queue, do not follow a normal dial plan
for the phone (don't send to user's voicemail). In stead, once timed out
(t|||60), send to Voicemail(u1000).
--
Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com
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Re: [asterisk-users] Dynamically Adding A Context
Seriously? You want serious! You can't handle the serious! I would assume that editing a file and refreshing a system by means of a program or self intervention which causes no interruption in service could be concidered dynamic. How does asterisk realtime handle this thats so radically different that it can be the only true dynamic method of doing this. BTW.. Did you figure it out yet? On 1/31/07, j [EMAIL PROTECTED] wrote: Seriously man. I don't want to be testy here, but what part of *dynamic* didn't you understand? Adding a context to a flat file and reloading the server is NOT dynamic. And, as I explained in a previous post, realtime is not a solution I can use for this issue because I'm updating proxy software that uses the AMI so realtime is not an option. For everyone else; Thanks for trying to take a stab at this. It seems there simply is no way to do it. Perhaps I'll submit a patch to digium so at least we have this simple functionality in the future... j On Tue, 2007-01-30 at 13:17 -0900, Shane Spencer wrote: Reload.. Reload.. Reload.. On 1/30/07, Shane Spencer [EMAIL PROTECTED] wrote: Realtime.. Realtime.. Realtime.. On 1/30/07, j [EMAIL PROTECTED] wrote: On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote: On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote: In order to do this, I have to add a couple quick extensions to the dial plan dynamically, so I have to be able to add my own context. from API use Command to run the CLI command add extension But you can only add to an existing context with that. Yes exactly. I tried the 'add extension' command. With *and* without the 'replace' argument, if the context does not already exist the command gives an error ;( ___ --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
[asterisk-users] Polycom IP 501+India
Hi Friends, This is Chandra from India. I have installed and configured Asterisk in our company. I want to provide Polycom IP 501 model phones to our employees. I am unable to find the dealer for these phones in India. Where can I buy these phones in India? If anybody knows, please tell me the dealer address or phone number. This is very urgent. Looking forward to your response. Thank you. Regards, Chandra. - Everyone is raving about the all-new Yahoo! Mail beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Status
Rob Schall wrote:
Hello all,
Lee recommended QUEUESTATUS, but that seems to return if anyone is in a
specific queue, and not if the current call came from a queue. I
probably just misunderstand how it all works. :)
Thanks all!
Rob
Hi Rob,
Remember that I am pretty new to Asterisk myself. ;)
I'm not sure how you have setup your queues, but with mine, the caller
doesn't go to the agent's voicemail (is it even supposed to?) if the
queue times out. If the queue times out for me, it will go to the next
line in the dialplan (or to the context specified if you allow them to
exit out).
So my suggestion was to see if QUEUESTATUS was set when the call kicks
out of the queue or into whatever context/extension in question, in a
way that could tell you if the call was coming out of a queue.
For my dialplans, I dedicate a single context for entering a queue like
this snippet from my dialplan:
[support_afterhours]
exten=s,1,Answer()
exten=s,2,Set(LAST_MENU_REACHED=support_afterhours)
exten=s,3,Playback(custom/support_reminder)
exten=s,4,Background(custom/support_afterhours)
exten=1,1,Queue(support,t|||60)
exten=1,2,Macro(DialExtenNoVM,112|60|tm)
Because this I already know that extension 1,1 is for entering the
queue, we can assume that if the call gets to 1,2 then there was a timeout.
If the context or extension is not setup so that you can assume a
timeout, then I was suggesting that you checkout the QUEUESTATUS variable:
From the wiki:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
The application sets the following channel variable upon completion:
QUEUESTATUS. The status of the call as a text string, one of TIMEOUT |
FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL
I just wasn't sure (haven't tested it) if the QUEUESTATUS var is
initially set as a null string if Queue() has not been called yet for
that channel. This would allow you to determine if the call was coming
out of a queue fairly easily:
exten=s,1,GotoIf($[${QUEUESTATUS} != ],?2:3)
exten=s,2,Macro(MyMacroToHandleQueueTimeouts)
exten=s,3,Macro(MyNormalDialplanLogicMacro)
Of course, it could be that I completely misunderstood what you are
trying to do ;)
--
Warm Regards,
Lee
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[asterisk-users] jastAGI
Hello I was wondering if anybody has some practical experience with JastAGI to share with me? thanks for all the 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] Queue Status
That's an interesting idea. Do you know if its possible to just check and see where the call came from. So if it came from the queue, do one thing, otherwise, do another? Rob Joe Dennick wrote: In the queues that I've established, I've assigned a different number to queue-agents than their normal extension. If their extension is 2120, their roll-over (second extension) would be 3120 and their queue-agent-id would be 4120. That way I can assign a different dial-plan for 4120 that doesn't include voicemail (or if it does, it's the queue's voicemail rather than the individual agent's voicemail). Hope that helps. Rob Schall wrote: Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk? If this call is coming from a queue, do not follow a normal dial plan for the phone (don't send to user's voicemail). In stead, once timed out (t|||60), send to Voicemail(u1000). Lee recommended QUEUESTATUS, but that seems to return if anyone is in a specific queue, and not if the current call came from a queue. I probably just misunderstand how it all works. :) Thanks all! Rob - I would recommend that you download the following tool and play with it (if you have a windows box): http://www.datatrakpos.com/pos/datatalk/Default.aspx Check out the Visual Menu Builder included. There is a Queue widget included. Try playing around with that and building the project and inspecting the resulting script that the program generates. This should give you a better idea of how to do what you are trying to do, I think. Also, check out the channel variable QUEUESTATUS. See if it's set to determine whether the call is coming from a queue. I'm not sure if it's normally blank or has a default value so you may want to check that out too. ___ --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] Toll-free dialing via PRI problem
This is a common issue with large inbound call center operations. They like to cheat. They actually start sending prompts to the caller without actually signalling their carrier that they have answered the line. Typically they do not answer until a phone is ringing or you are in a queue. I do believe this is illegal per the FCC. From asterisk, you do not hear anything other than ringing as it does not cut the audio path through until it receives the answer from the far end, hence the steady ringing. Forgive my impetuousness, but what's the differentiator that allows calls to work from my legacy PBX and cell phone that precludes the Asterisk from working as the others? Just askin' is all... ;-) Stefano ___ --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] put Agi script in queue
Asterisk version 1.4 Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]) The optional AGI parameter will setup an AGI script to be executed on the calling party's channel once they are connected to a queue member. Ceará Kind of - you could link that to the Local/xxx channel called for agents, or you could fork the dialplan and on one branch send the user to the queue and on the other one run the AGI. l. On Mon, 29 Jan 2007 15:55:21 +0100, nik600 [EMAIL PROTECTED] wrote: Hi everyone dou you know if is possible to put an Agi script in a queue? For Example 1 - Caller joins the queue 2 - Agi script starts ... ... Agi script ends 3 - Hangup. Is it possible? thanks -- 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] Problem with Voipjet ...
also trixbox stop registering randomly on all versions.. confirmed with over 200 client accounts over here... all using trxibox.. asterisk vanilla is ok On 1/31/07, Alejandro Lengua [EMAIL PROTECTED] wrote: How many simultaneous calls per account are you sending ? On 1/31/07, Peter Halliday [EMAIL PROTECTED] wrote: That's interesting I use Voipjet cheap lines and I don't have a problem at all. Peter On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Hello, we have this problem with Trixbox 1.23 I have created an outgoing route where the 1st line has Voipjet and the 2nd an 3rd have voipcheap accounts. The problem is that at certain moments, when we call all the calls go through the voipcheap SIP accounts SIP, whose quality are not only not good enough but also consume a lot of bandwidth. The error message that returns Voipjet to Asterisk is that all circuits busy. What I asume from this? Thanks in advance ___ ___ --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 -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --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] Hi Honies! I'm home!
Your dinner's in the oven. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Status
Lee Jenkins wrote:
Rob Schall wrote:
Hello all,
Lee recommended QUEUESTATUS, but that seems to return if anyone is in a
specific queue, and not if the current call came from a queue. I
probably just misunderstand how it all works. :)
Thanks all!
Rob
Hi Rob,
Remember that I am pretty new to Asterisk myself. ;)
I'm not sure how you have setup your queues, but with mine, the caller
doesn't go to the agent's voicemail (is it even supposed to?) if the
queue times out. If the queue times out for me, it will go to the next
line in the dialplan (or to the context specified if you allow them to
exit out).
So my suggestion was to see if QUEUESTATUS was set when the call kicks
out of the queue or into whatever context/extension in question, in a
way that could tell you if the call was coming out of a queue.
For my dialplans, I dedicate a single context for entering a queue like
this snippet from my dialplan:
[support_afterhours]
exten=s,1,Answer()
exten=s,2,Set(LAST_MENU_REACHED=support_afterhours)
exten=s,3,Playback(custom/support_reminder)
exten=s,4,Background(custom/support_afterhours)
exten=1,1,Queue(support,t|||60)
exten=1,2,Macro(DialExtenNoVM,112|60|tm)
Because this I already know that extension 1,1 is for entering the
queue, we can assume that if the call gets to 1,2 then there was a timeout.
If the context or extension is not setup so that you can assume a
timeout, then I was suggesting that you checkout the QUEUESTATUS variable:
From the wiki:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
The application sets the following channel variable upon completion:
QUEUESTATUS. The status of the call as a text string, one of TIMEOUT |
FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL
I just wasn't sure (haven't tested it) if the QUEUESTATUS var is
initially set as a null string if Queue() has not been called yet for
that channel. This would allow you to determine if the call was coming
out of a queue fairly easily:
exten=s,1,GotoIf($[${QUEUESTATUS} != ],?2:3)
exten=s,2,Macro(MyMacroToHandleQueueTimeouts)
exten=s,3,Macro(MyNormalDialplanLogicMacro)
Of course, it could be that I completely misunderstood what you are
trying to do ;)
I forgot that you could also just set a channel variable before sending
the caller to a queue. Then check for that variable in other parts of
the script.
--
Warm Regards,
Lee
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Re: [asterisk-users] To 1.4 or not
I thought it was If it ain't broke fix it till it is!? C F wrote: Change log can help you a lot. I would stick to my grandmothers advice, if it aint broken don't fix it. On 1/14/07, Yuan LIU [EMAIL PROTECTED] wrote: I don't have a particular reason to upgrade, but I'm installing a new box, so I have the opportunity to go 1.4. On the other hand, I'm not familiar with 1.4, and relatively new to Asterisk. So instead of trying to keep up with two different versions, I want to tie my handful of boxes to one, before any of them grow too complex. Is there a document about the main motivations to upgrade? From your practice, what are your primary reasons? Thank you in advance. Yuan Liu ___ --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 dual contexts stupidity
At 09:20 AM 1/31/2007, you wrote: I was once working on tracking down a particularly elusive bug in one of our products and put a small piece of code showing a message when testing a value that said Sh*t still doesn't work if a certain value was true. I guess you already know that I forgot to remove this little tidbit before distributing the update. I've always been more inclined to use horse, cow, fish and truck. Confuses the heck out of the users but you never get calls from the CEO of a Fortune 500 wondering why your application is creating files called sh??.tmp on his secretary's hard drive. A friends boss at a Big 8 accounting firm got that call, not a good thing! Ira ___ --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 sip peer/user matching methodsforauthentication backwards?
Anyone found a solution to this problem? Remi Damon Estep wrote: I have considered opening a bug report on this, but wanted to get some feedback and make sure I am not missing something in the way of a simple work around. What is the scenario in which this impacts your implementation? Ours is the desire to use the same realtime SIP database for many asterisk servers, and route the call based on a “home server” value in the realtime database. The problem is that a call routed form one server to another will not complete because the originating server is not trusted as it should be by IP address, rather the SIP UA that initiated the call is expected to authenticate on the destination server, which is ridiculous. All methods of allowing un-authenticated SIP peering (host=, insecure=) are broken as soon as the caller name portion of the “from” header URI is present on the called parties server. I can not think of why it would break something different to reverse the evaluation order. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Doug Meredith *Sent:* Thursday, January 04, 2007 10:23 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [asterisk-users] asterisk sip peer/user matching methodsforauthentication backwards? Hi, I too have found this matching to be frustrating. I would like it to behave as you describe. Doug -- Doug Meredith 506-854-7997 ext. 801 *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Damon Estep *Sent:* Thursday, January 04, 2007 1:50 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] asterisk sip peer/user matching methods forauthentication backwards? Take an example where there is two sip users defined in sip.conf as follows; [peer1] Host=192.168.1.1 … [peer2] Host=dynamic Secret=password … [Peer3] Config not relevant … The intention is to accept calls from peer1 without authentication (ip address authentication only), but require authentication from peer2 If by chance a SIP invite comes “From” [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (where the name peer2 on the calling server coincidentally matches a defined sip user on the called asterisk server) “To” [EMAIL PROTECTED], Asterisk will attempt to authenticate the caller “peer2” rather than accepting the call based on the fact that it came from a trusted Ip address defined for peer1. Since peer1 is trusted it is not sending credentials and will have its invite rejected with a 407 “proxy authentication required” when it fails to authenticate as “peer2”. This logic seems backwards to me, the IP address should be matched first, and if there is no statically defined user with that IP address the username should be matched next. This would insure that all calls from the trusted IP address are accepted regardless of whether there is coincidently a SIP user with a matching name defined on the target asterisk server. So rather than looking for a match in this order; 1. name portion of “From” URI in the invite (“host” in the URI [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]). 2. ip address statically assigne for a user it should look in this order; 1. statically defined sip user ip addresses 2. name portion of the “From” URI Can anyone shed any light on this, or suggest a workaround so 407’s are not sent if the invite “from” header happens to have the same name portion of the URI as a defined sip user on the target asterisk server ? ___ --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: [asterisk-dev] Dynamically Adding A Context
On Wed, 2007-01-31 at 12:14 -0500, Lee Jenkins wrote: yusuf wrote: j wrote: Greetings! I've searched far and wide for an answer and have gotten no where, so I was hoping one of you guys might have the answer; Is it possible to dynamically add a context to the dialplan? You can add extensions via the CLI, however if the context doesn't exist I get an error message instead of it creating the context for me. Any method will do, AGI, AMI, CLI... I just need a solution :) Yusuf, I was just curious what kind of context do you need to add? The reason I ask is maybe you could use a custom AGI script to simulate the same steps that would occur in a dynamically added context? You could for instance, create an AGI that reads from a dynamically created text file template or flat script? Just a thought... Good thought. Thing is, what exactly would the agi script do? There's no agi command to add a context either. There's no way to simulate a dynamically added context mainly because, well, there's nothing to simulate :) ___ --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] Dynamically Adding A Context
On Tue, 2007-01-30 at 20:17 -0800, Ira wrote: At 02:17 PM 1/30/2007, you wrote: Yes exactly. I tried the 'add extension' command. With *and* without the 'replace' argument, if the context does not already exist the command gives an error ;( You could create a set of empty extensions to use and re-use as needed. It's one of those tasks that made you wish array() had something to do with arrays!!! Ira Thanks for the thought, unfortunately this won't work for what I'm trying to do. The software I'm building is being used on arbitrary systems all over the world, so I can't count on real time being there and I refuse to make it edit the extensions.conf and reload extensions (how would you feel if you used some proxy software and it started messing with your extensions.conf file??)... j ___ --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] Dynamically Adding A Context
On Wed, 2007-01-31 at 09:28 -0900, Shane Spencer wrote: Seriously? You want serious! You can't handle the serious! Heh. Sorry man, it's been a bad day :( I would assume that editing a file and refreshing a system by means of a program or self intervention which causes no interruption in service could be concidered dynamic. I suppose the interpretation of dynamic can be somewhat subjective, however I tend to think of a text file as static. Dynamic, IMO, would be more along the lines of manipulating the configuration of the machine on-the-fly directly. How does asterisk realtime handle this thats so radically different that it can be the only true dynamic method of doing this. The problem with realtime is that not everyone uses it. I'm not trying to do something specific to my own configuration. I'm updating proxy software that's used in arbitrary environments, so I can only count on stuff that everyone has. BTW.. Did you figure it out yet? Unfortunately, it doesn't seem as though there is a way. I had to resort to making the users of the software map dynamic agent channels (i.e. Local/[EMAIL PROTECTED]) to actual device channels (i.e. SIP/200) in the users configuration file for every single user. I'm sure I'll hear some grumbling from some of the users I've spoken with in the past with very large call centers :( It would have been really cool if I could have dynamically called and traced the dynamic agents to their actual devices and removed the headache from the admin .. but .. oh well :( On 1/31/07, j [EMAIL PROTECTED] wrote: Seriously man. I don't want to be testy here, but what part of *dynamic* didn't you understand? Adding a context to a flat file and reloading the server is NOT dynamic. And, as I explained in a previous post, realtime is not a solution I can use for this issue because I'm updating proxy software that uses the AMI so realtime is not an option. For everyone else; Thanks for trying to take a stab at this. It seems there simply is no way to do it. Perhaps I'll submit a patch to digium so at least we have this simple functionality in the future... j On Tue, 2007-01-30 at 13:17 -0900, Shane Spencer wrote: Reload.. Reload.. Reload.. On 1/30/07, Shane Spencer [EMAIL PROTECTED] wrote: Realtime.. Realtime.. Realtime.. On 1/30/07, j [EMAIL PROTECTED] wrote: On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote: On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote: In order to do this, I have to add a couple quick extensions to the dial plan dynamically, so I have to be able to add my own context. from API use Command to run the CLI command add extension But you can only add to an existing context with that. Yes exactly. I tried the 'add extension' command. With *and* without the 'replace' argument, if the context does not already exist the command gives an error ;( ___ --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] Dynamically Adding A Context
Cool. My first attempt would have been to find out how to use asterisk variables in the dialplan since I can set those like crazy mad via an AGI. Then i would have cried and become horribly demotivated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with semaphores
From:Mitch Thompson [EMAIL PROTECTED]I'm looking for some help from any Asterisk "heavy" who might be doing something similar to what I'm trying to do...Background:I work for a research lab, testing telephony products and tools.Historically, we used Ameritec Crescendos and Fortissimos to act as load generators and call "sinks" when testing equipment.However, the equipment we are testing gets more and more complex, and the scripted scenarios the Ameritecs give have become a limiting factor for testing.Therefore, Asterisk was chosen as a possible solution (we're a cheap lab).
Mitch,
I had exposure to both Ameritec and Hammer, and see how Ameritec could be limiting. But using a PBX as a test tool doesn't sound very sound even for a cheap lab, especially for load test. Race condition is just one side of the problem. You also have to spend a lot of time programming the PBX to do what test tools are designed to do. Have you looked into Sprient? They boast the highest density per $in PSTN land but I don't know the scripting capability.
Back to your condition. You can replace cat with test in priority 1to reduce time consumed by the first system call by half, thus theoretically speed up branching to priority 102. But the bottleneckis likely in Asterisk's branch codes. Hence even if the system call takes no time, even if you store stuff in memory,you are still going to run into race conditions.
Yuan Liu
I've been learning Asterisk as I go, but I've learned a lot.Here's the basic scenario:We are using an Asterisk (AAH 2.8, specifically) to sink calls.I do this by taking the ${EXTEN} and breaking it down by sections until I get to the last 4 digits (i.e., 2105551212).Once I get to the 4-digit extension, I am trying to set a flag, or semaphore, to do Busy/Idle testing.Here is my extensions_custom.conf fragment:[SATX_555_Extensions]exten = 1212,1,System(cat /tmp/{orig_num}); ${orig_num} is set at the beginning of [from-trunk-custom] to the full dialed digits in ${EXTEN}, before I break it down.exten = 1212,n,Busy(); if the file exists,
someone else has already called this number, return busyexten = 1212,102,System(echo ${UNIQUEID} /tmp/${orig_num}) ; basically, create a file in /tmp whose name is the full number from the beginning.In this case, the full
; filename would be /tmp/2105551212.I don't really care about the contents, though.exten = 1212,103, Goto(Idle,1) ; from here, we jump to a new extension called Idle, where we do a Random to decide whether to simulate no one home (ring no answer) or; we send ring
for about 10 seconds, then Answer() and play some .wav files, then hangup.The last thing we do in either case is to delete; the /tmp/${orig_num} file.The above code works very well at low call volumes.However, I'm running into race conditions at high call volumes where several calls are getting through the test in priority 1 before the file is created in priority 102 (n+101).I've tried to implement semaphores by using both local and global variables, but it
doesn't seem to work.My ultimate question:Is anyone doing something similar, and what did you do to implement the busy/idle.I appreciate any help anyone can offer.Mitch Thompson
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Re: [asterisk-users] Dynamically Adding A Context
lol. On Wed, 2007-01-31 at 13:00 -0900, Shane Spencer wrote: Cool. My first attempt would have been to find out how to use asterisk variables in the dialplan since I can set those like crazy mad via an AGI. Then i would have cried and become horribly demotivated. ___ --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: [asterisk-dev] Dynamically Adding A Context
From:j [EMAIL PROTECTED]On Wed, 2007-01-31 at 12:14 -0500, Lee Jenkins wrote: I've searched far and wide for an answer and have gotten no where, so I was hoping one of you guys might have the answer; Is it possible to dynamically add a context to the dialplan? You can add extensions via the CLI, however if the context doesn't exist I get an error message instead of it creating the context for me. Any method will do, AGI, AMI, CLI... I just need a solution :) Yusuf, I was just curious what kind of context do you need to add?The reason I ask is maybe you could use a custom AGI script to simulate the same steps that would occur in a dynamically added context? You could for instance, create an AGI that reads from a dynamically created text file template or flat script? Just a thought... Good thought. Thing is, what exactly would the agi script do? There'sno agi command to add a context either. What Lee suggested is to have the AGI script to actually parse, insert a new context in extensions.conf, or deleting from it, then reload extensions.conf. This would at least achieve what you wanted to do. Yuan Liu There's no way to simulate a dynamically added context mainly because,well, there's nothing to simulate :) ___ --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: [asterisk-dev] Dynamically Adding A Context
Reload.. Reload.. Reload..! /me ducks ___ --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: [asterisk-dev] Dynamically Adding A Context
Shane Spencer wrote: Reload.. Reload.. Reload..! LOL. -- Warm Regards, Lee ___ --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] how to get the status of failed call files
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't
used) and channel_status doesn't seem to be any good.
thanks in advance.
--
- Rich Doughty
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Re: [asterisk-users] Re: [asterisk-dev] Dynamically Adding A Context
On 01/02/07, Yuan LIU [EMAIL PROTECTED] wrote: What Lee suggested is to have the AGI script to actually parse, insert a new context in extensions.conf, or deleting from it, then reload extensions.conf. This would at least achieve what you wanted to do. Or alternatively, to avoid complete disaster, why not have extensions.conf include another file (#include somefile.conf) and edit that one with your script? I've done that before (although I was actually recreating the entire file each time by populating from an external database). Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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] how to get the status of failed call files
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't
used) and channel_status doesn't seem to be any good.
thanks in advance.
the event you received for OriginateFailure has a 'Reason: ' code.
that code breaks down as
0 = UNKNOWN FAILURE or DISCONNECT
3 = AST_CONTROL_RINGING (no answer)
5 = AST_CONTROL_BUSY
1 = AST_CONTROL_HANGUP
8 = AST_CONTROL_CONGESTION
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Re: [asterisk-users] Re: [asterisk-dev] Dynamically Adding A Context
Hahaha,, I think thats a freaking SWEET suggestion :) On 1/31/07, Andrew Furey [EMAIL PROTECTED] wrote: On 01/02/07, Yuan LIU [EMAIL PROTECTED] wrote: What Lee suggested is to have the AGI script to actually parse, insert a new context in extensions.conf, or deleting from it, then reload extensions.conf. This would at least achieve what you wanted to do. Or alternatively, to avoid complete disaster, why not have extensions.conf include another file (#include somefile.conf) and edit that one with your script? I've done that before (although I was actually recreating the entire file each time by populating from an external database). Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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] Toll-free dialing via PRI problem
Jerry Jones wrote: From asterisk, you do not hear anything other than ringing as it does not cut the audio path through until it receives the answer from the far end, hence the steady ringing. So instead of Dial(Zap/g1/1800xxx,,r) just do Dial(Zap/g1/1800xxx,,) so early audio can make it through. Unless there's more to the puzzle? ___ --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] How would you compare feature set to a Metaswitch?
OK I need some help. Looking for comparisons for a large customer wishing to provide voip service over a region. We are up against Metaswitch who is claiming they can do anything Asterisk can do. I do not have too much information on Metaswitch so am looking for any information, preferably real world experience on how Asterisk and Metaswitch would compare side by side. Thanks in advance. Jerry ___ --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] Which Java FastAGI implementation has the most market share?
When I was looking for a Java FastAGI interface for Asterisk I came across asterisk-java first and didn't realize there was more than one out there. It seems to work fine and I've got my first project working with it, but I was wondering which Java FastAGI implementation is the most popular and how they compare against each other. So I'm aware of: asterisk-java JastAGI OrderlyCalls Any comments on who the front runner is and why? 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
[asterisk-users] FreePBX/Debian Aborts Call While Connecting
I used the FreePBX on Debian HowTo at
http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles
to initiate calls to my SIP carrier. They get my registration, but they
see that my call is interrupted before they can complete the connection.
My Asterisk log shows that the call times out after the time (45s)
specified in my dialplan Dial() command. What is wrong?
[from /var/log/asterisk/full]:
Jan 30 23:40:35 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command
'Command'
Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command
'Command'
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Scheduled a registration timeout
for 66.153.22.16 id #17818
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 606: Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 606: Match
Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 607: Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 607: Match
Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Registration successful
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Cancelling timeout 17818
Jan 30 23:41:16 DEBUG[6245] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]'
Jan 30 23:41:30 VERBOSE[17267] logger.c: -- Attempting call on
Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 (Retry 1)
Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Executing
NoOp(Local/[EMAIL PROTECTED],2, Calling
SIP/[EMAIL PROTECTED]) in new stack
Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Executing
Dial(Local/[EMAIL PROTECTED],2, SIP/[EMAIL PROTECTED]|45|
M(say-call-2-digits^17182335097)g) in new stack
Jan 30 23:41:30 DEBUG[17269] chan_sip.c: Setting NAT on RTP to 0
Jan 30 23:41:30 DEBUG[17269] chan_sip.c: Outgoing Call for 16467508273
Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Called
[EMAIL PROTECTED]
Jan 30 23:41:30 DEBUG[6245] chan_sip.c: Acked pending invite 102
Jan 30 23:41:30 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jan 30 23:41:30 DEBUG[6245] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 103: Found
Jan 30 23:41:35 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jan 30 23:42:15 VERBOSE[17269] logger.c: -- Nobody picked up in
45000 ms
Jan 30 23:42:15 DEBUG[17269] chan_sip.c:
update_call_counter(16467508273) - decrement call limit counter
Jan 30 23:42:15 DEBUG[17269] chan_sip.c: Acked pending invite 103
Jan 30 23:42:15 DEBUG[17269] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Match
Found
Jan 30 23:42:15 DEBUG[17269] app_dial.c: Exiting with
DIALSTATUS=NOANSWER.
Jan 30 23:42:15 VERBOSE[17269] logger.c: -- Executing
NoOp(Local/[EMAIL PROTECTED],2, Done dialing from) in new
stack
Jan 30 23:42:15 DEBUG[17267] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jan 30 23:42:15 DEBUG[17267] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2007-01-30
23:42:15','16467508273','16467508273','callTo','ext-jjp-out', 'Local/[EMAIL
PROTECTED],1','','Dial','Local/[EMAIL
PROTECTED]/n',0,0,'FAILED',3,'','1170196890.32')
Jan 30 23:42:15 DEBUG[17267] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jan 30 23:42:15 DEBUG[17267] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2007-01-30 23:42:15','','','s','default',
'**Unknown**','','','',0,0,'FAILED',3,'','1170196935.35')
Jan 30 23:42:15 NOTICE[17267] pbx_spool.c: Call failed to go through,
reason 0
Jan 30 23:42:15 DEBUG[17269] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jan 30 23:42:15 DEBUG[17269] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2007-01-30
23:41:30','16467508273','16467508273','callFrom','ext-jjp-out', 'Local/[EMAIL
PROTECTED],2','SIP/tu3961-08196340','NoOp','Done dialing from',45,0,'NO
ANSWER',3,'','1170196890.33')
Jan 30 23:42:15 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Match
Not Found
Jan 30 23:42:15 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Match
Found
Jan 30 23:42:29 DEBUG[6245] chan_sip.c: Scheduled a registration timeout
for 66.153.22.16 id #17831
Jan 30
[asterisk-users] kewlstart disconnect threshold
Hi, folks: Can the loop drop detection threshold (normally defined in milliseconds) be set on the Digium TDM-400 cards? Most PBXs let you set this value. -Stephen- ___ --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] Toll-free dialing via PRI problem
Sheepishly, that was the magic bullet. Thanks Trevor!! Tim Trevor Peirce [EMAIL PROTECTED] wrote: Jerry Jones wrote: From asterisk, you do not hear anything other than ringing as it does not cut the audio path through until it receives the answer from the far end, hence the steady ringing. So instead of Dial(Zap/g1/1800xxx,,r) just do Dial(Zap/g1/1800xxx,,) so early audio can make it through. Unless there's more to the puzzle? ___ --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] no lights on TE405P, but shows up in lspci, modules loaded
I pulled a working TE405P from one box and put it in another box. I compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no lights on the card come on. I do an lspci and the card shows up there. I ran ztcfg -vv and got the error message Unable to open master device '/dev/zap/ctl' so I followed the instructions in README.udev the error message went away, but now when I run ztcfg I just get 0 channels configured Thanks! Wayne ___ --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] kewlstart disconnect threshold
Stephen Bosch wrote: Hi, folks: Can the loop drop detection threshold (normally defined in milliseconds) be set on the Digium TDM-400 cards? Most PBXs let you set this value. Good question. Anyone knows if the TDM-400 actually detect loop drops? Leo ___ --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] no lights on TE405P, but shows up in lspci, modules loaded
Is udev running? On 1/31/07, Wayne Jensen [EMAIL PROTECTED] wrote: I pulled a working TE405P from one box and put it in another box. I compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no lights on the card come on. I do an lspci and the card shows up there. I ran ztcfg -vv and got the error message Unable to open master device '/dev/zap/ctl' so I followed the instructions in README.udev the error message went away, but now when I run ztcfg I just get 0 channels configured Thanks! Wayne ___ --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] no lights on TE405P, but shows up in lspci, modules loaded
yes On 1/31/07, C F [EMAIL PROTECTED] wrote: Is udev running? On 1/31/07, Wayne Jensen [EMAIL PROTECTED] wrote: I pulled a working TE405P from one box and put it in another box. I compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no lights on the card come on. I do an lspci and the card shows up there. I ran ztcfg -vv and got the error message Unable to open master device '/dev/zap/ctl' so I followed the instructions in README.udev the error message went away, but now when I run ztcfg I just get 0 channels configured Thanks! Wayne ___ --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] no lights on TE405P, but shows up in lspci, modules loaded
On Wed, Jan 31, 2007 at 08:40:59PM -0700, Wayne Jensen wrote: I pulled a working TE405P from one box and put it in another box. I compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no lights on the card come on. I do an lspci and the card shows up there. I ran ztcfg -vv and got the error message Unable to open master device '/dev/zap/ctl' so I followed the instructions in README.udev the error message went away, but now when I run ztcfg I just get 0 channels configured Do you see relevant entries under /sys/class/zaptel ? Later on, do you have /dev/zap/ctl ? If you run ztcfg later, do you still get the same error? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kewlstart disconnect threshold
On Wed, Jan 31, 2007 at 07:57:54PM -0700, Stephen Bosch wrote: Hi, folks: Can the loop drop detection threshold (normally defined in milliseconds) be set on the Digium TDM-400 cards? Most PBXs let you set this value. What exactly do you need it for? On the FXO module (detecting) or on the FXS module (generating)? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP 501+India
If you already havent seen this: http://dir.indiamart.com/impcat/video-telephone.html cheerz - Ben. Crazy Boy wrote: Hi Friends, This is Chandra from India. I have installed and configured Asterisk in our company. I want to provide Polycom IP 501 model phones to our employees. I am unable to find the dealer for these phones in India. Where can I buy these phones in India? If anybody knows, please tell me the dealer address or phone number. This is very urgent. Looking forward to your response. Thank you. Regards, Chandra. Everyone is raving about the all-new Yahoo! Mail beta. http://us.rd.yahoo.com/evt=45083/*http://advision.webevents.yahoo.com/mailbeta ___ --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 problem with the Future is that it keeps turning into the Present. *** EMAIL DISCLAIMER : *** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ** ___ --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] kewlstart disconnect threshold
Tzafrir, I'm assuming FXO module, since that's where one can usualy (on other PBXs) set it. On 1/31/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jan 31, 2007 at 07:57:54PM -0700, Stephen Bosch wrote: Hi, folks: Can the loop drop detection threshold (normally defined in milliseconds) be set on the Digium TDM-400 cards? Most PBXs let you set this value. What exactly do you need it for? On the FXO module (detecting) or on the FXS module (generating)? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with semaphores
Yuan LIU wrote:
From: /Mitch Thompson [EMAIL PROTECTED]/
I'm looking for some help from any Asterisk heavy who might be
doing something similar to what I'm trying to do...
Background:
I work for a research lab, testing telephony products and tools.
Historically, we used Ameritec Crescendos and Fortissimos to act as
load generators and call sinks when testing equipment. However,
the equipment we are testing gets more and more complex, and the
scripted scenarios the Ameritecs give have become a limiting factor
for testing. Therefore, Asterisk was chosen as a possible solution
(we're a cheap lab).
Mitch,
I had exposure to both Ameritec and Hammer, and see how Ameritec could
be limiting. But using a PBX as a test tool doesn't sound very sound
even for a cheap lab, especially for load test. Race condition is
just one side of the problem. You also have to spend a lot of time
programming the PBX to do what test tools are designed to do. Have
you looked into Sprient? They boast the highest density per $ in PSTN
land but I don't know the scripting capability.
Back to your condition. You can replace cat with test in priority
1 to reduce time consumed by the first system call by half, thus
theoretically speed up branching to priority 102. But the
bottleneck is likely in Asterisk's branch codes. Hence even if the
system call takes no time, even if you store stuff in memory, you are
still going to run into race conditions.
Yuan Liu
I've been learning Asterisk as I go, but I've learned a lot. Here's
the basic scenario:
We are using an Asterisk (AAH 2.8, specifically) to sink calls. I
do this by taking the ${EXTEN} and breaking it down by sections
until I get to the last 4 digits (i.e., 2105551212). Once I get to
the 4-digit extension, I am trying to set a flag, or semaphore, to
do Busy/Idle testing. Here is my extensions_custom.conf fragment:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at
the beginning of [from-trunk-custom] to the full dialed digits in
${EXTEN}, before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has already
called this number, return busy
exten = 1212,102,System(echo ${UNIQUEID} /tmp/${orig_num}) ;
basically, create a file in /tmp whose name is the full number from
the beginning. In this case, the full
; filename would be
/tmp/2105551212. I don't really care about the contents, though.
exten = 1212,103, Goto(Idle,1) ; from here, we jump to a new
extension called Idle, where we do a Random to decide whether to
simulate no one home (ring no answer) or
; we send ring
for about 10 seconds, then Answer() and play some .wav files, then
hangup. The last thing we do in either case is to delete
; the
/tmp/${orig_num} file.
The above code works very well at low call volumes. However, I'm
running into race conditions at high call volumes where several
calls are getting through the test in priority 1 before the file is
created in priority 102 (n+101).
I've tried to implement semaphores by using both local and global
variables, but it doesn't seem to work.
My ultimate question: Is anyone doing something similar, and what
did you do to implement the busy/idle.
I appreciate any help anyone can offer.
Mitch Thompson
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Thank you very much for the feedback. I'll pass it on to the engineer
(I'm the lowly tech who is doing the implementing.) One good deal is
that they're sending me to Asterisk Bootcamp sometime in the next few months
--
For unto you is born this day in the city of David a Saviour, which is
Christ the Lord. Luke 2:11
--
Read The Patriot It's Right -- It's Free
http://PatriotPost.US/subscribe/
--
Mitch Thompson, San Antonio, Texas//WB5UZG
Red Hat Certified Engineer
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Re: [asterisk-users] no lights on TE405P, but shows up in lspci, modules loaded
On 1/31/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jan 31, 2007 at 08:40:59PM -0700, Wayne Jensen wrote: I pulled a working TE405P from one box and put it in another box. I compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no lights on the card come on. I do an lspci and the card shows up there. I ran ztcfg -vv and got the error message Unable to open master device '/dev/zap/ctl' so I followed the instructions in README.udev the error message went away, but now when I run ztcfg I just get 0 channels configured Do you see relevant entries under /sys/class/zaptel ? Later on, do you have /dev/zap/ctl ? If you run ztcfg later, do you still get the same error? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Nevermind. I scrapped Ubuntu, installed Debian, and all is well. ___ --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] Regarding Call Queue
Hi My agents are logged into the queue through the UI but when ever I run show queue on the console, it shows Members Agent 6001 (Unavailable) Agent 6002 (Unavailable) NO callers I have followed all the steps in Queue creation. And whenever I place a call at Queue, I hear a hold tone. Instead if I call directly at my agents extension then the phone rings. How to overcome this problem. With Regards Manish Gupta -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Wednesday, January 31, 2007 8:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Regarding Call Queue Is your agent logged into that queue to receive the calls? You can typically say show queues to list all queues and see who is not in use vs unavailable. If they are all unavailable, are you getting a successful Agent Logged In message when you log that guy in? Rob Manish Gupta02 wrote: Hi I recently installed AsteriskNOW and I am trying to use its Call queue feature. But after configuring the Queue whenever I place a call, no phone in my Queue list rings. I am not able to overcome this problem. I am using Snom360 as my softphone. Please help me In this regard. With Regards Manish Gupta CAUTION - Disclaimer * This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS End of Disclaimer INFOSYS*** ___ --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] Fax from PAP2 through a zap channel to PSTN
Hello all, Can I send fax from PAP2 through a zap channel to PSTN? I have tried but it is not successful. Thank you for your help! Lai ___ --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] Softphone for Palm
Anyone know of a softphone for the Palm OS ? Thanks. Dovid___ --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] strange caller display
Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows asterisk when I make a call to the receiver. I wonder why asterisk shows in the display as I haven't set any word - asterisk in any configuration file. How to remove that word from the receive end if it is a default word? Below is the log dump from ngrep. There is no asterisk in the from header except the option message. I wonder why asterisk will be shown in the receiver end's screen. ango U 10.0.0.25:2750 - 10.201.0.224:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 10.0.0.25:2750;branch=z9hG4bK-d87543-5d65ca22ac139c29-1--d87543-;rport. Max-Forwards: 70. Contact: sip:[EMAIL PROTECTED]:2750. To: 85236418505sip:[EMAIL PROTECTED]. From: angry boysip:[EMAIL PROTECTED];tag=b842555d. Call-ID: MWE0YWUzYzBmOTQ2NGUyYTE2OTAxYTRlYTk4ODAxOTA.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 1008b stamp 36844. Content-Length: 449. . v=0. o=- 6 2 IN IP4 10.0.0.25. s=CounterPath eyeBeam 1.5. c=IN IP4 10.0.0.25. t=0 0. m=audio 4148 RTP/AVP 98 18 3 101. a=alt:1 3 : 6ceGNvpQ aNAT7Mk6 10.0.0.25 4148. a=alt:2 2 : cu+cL3mB rdqEXGtX 192.168.132.1 4148. a=alt:3 1 : uoim9Hbs Eiu4Y4zw 192.168.80.1 4148. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:98 iLBC/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=x-rtp-session-id:E06A42E19E7244AFBF10DCAF883B488B. # U 10.201.0.224:5060 - 10.0.0.25:2750 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.0.0.25:2750;branch=z9hG4bK-d87543-5d65ca22ac139c29-1--d87543-;received=10.0.0.25;rport=2750. From: angry boysip:[EMAIL PROTECTED];tag=b842555d. To: 85236418505sip:[EMAIL PROTECTED];tag=as49ada1cc. Call-ID: MWE0YWUzYzBmOTQ2NGUyYTE2OTAxYTRlYTk4ODAxOTA.. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=08efb083. Content-Length: 0. . # U 10.0.0.25:2750 - 10.201.0.224:5060 ACK sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 10.0.0.25:2750;branch=z9hG4bK-d87543-5d65ca22ac139c29-1--d87543-;rport. To: 85236418505sip:[EMAIL PROTECTED];tag=as49ada1cc. From: angry boysip:[EMAIL PROTECTED];tag=b842555d. Call-ID: MWE0YWUzYzBmOTQ2NGUyYTE2OTAxYTRlYTk4ODAxOTA.. CSeq: 1 ACK. Content-Length: 0. . # U 10.0.0.25:2750 - 10.201.0.224:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 10.0.0.25:2750;branch=z9hG4bK-d87543-f3250e403844c711-1--d87543-;rport. Max-Forwards: 70. Contact: sip:[EMAIL PROTECTED]:2750. To: 85236418505sip:[EMAIL PROTECTED]. From: angry boysip:[EMAIL PROTECTED];tag=b842555d. Call-ID: MWE0YWUzYzBmOTQ2NGUyYTE2OTAxYTRlYTk4ODAxOTA.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username=9000220002,realm=asterisk,nonce=08efb083,uri=sip:[EMAIL PROTECTED],response=b50237df83b408c2e7898e0da9153bef,algorithm=MD5. User-Agent: eyeBeam release 1008b stamp 36844. Content-Length: 449. . v=0. o=- 6 2 IN IP4 10.0.0.25. s=CounterPath eyeBeam 1.5. c=IN IP4 10.0.0.25. t=0 0. m=audio 4148 RTP/AVP 98 18 3 101. a=alt:1 3 : 6ceGNvpQ aNAT7Mk6 10.0.0.25 4148. a=alt:2 2 : cu+cL3mB rdqEXGtX 192.168.132.1 4148. a=alt:3 1 : uoim9Hbs Eiu4Y4zw 192.168.80.1 4148. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:98 iLBC/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=x-rtp-session-id:E06A42E19E7244AFBF10DCAF883B488B. # U 10.201.0.224:5060 - 10.0.0.25:2750 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.25:2750;branch=z9hG4bK-d87543-f3250e403844c711-1--d87543-;received=10.0.0.25;rport=2750. From: angry boysip:[EMAIL PROTECTED];tag=b842555d. To: 85236418505sip:[EMAIL PROTECTED]. Call-ID: MWE0YWUzYzBmOTQ2NGUyYTE2OTAxYTRlYTk4ODAxOTA.. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: sip:[EMAIL PROTECTED]. Content-Length: 0. . # U 10.201.0.224:5060 - 10.0.0.25:2750 OPTIONS sip:[EMAIL PROTECTED]:2750;rinstance=f136277835976893 SIP/2.0. Via: SIP/2.0/UDP 10.201.0.224:5060;branch=z9hG4bK3069abdf;rport. From: asterisk sip:[EMAIL PROTECTED];tag=as77042273. To: sip:[EMAIL PROTECTED]:2750;rinstance=f136277835976893. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Thu, 01 Feb 2007 06:33:00 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Length: 0. . # U 10.0.0.25:2750 - 10.201.0.224:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.201.0.224:5060;branch=z9hG4bK3069abdf;rport=5060. Contact: sip:10.0.0.25:2750. To: sip:[EMAIL
Re: [asterisk-users] FreePBX/Debian Aborts Call While Connecting
Yeah, your waittime parameter in your call file is set to 45 seconds.
db
On Wed, 2007-01-31 at 21:52 -0500, Matthew Rubenstein wrote:
I used the FreePBX on Debian HowTo at
http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles
to initiate calls to my SIP carrier. They get my registration, but they
see that my call is interrupted before they can complete the connection.
My Asterisk log shows that the call times out after the time (45s)
specified in my dialplan Dial() command. What is wrong?
[from /var/log/asterisk/full]:
Jan 30 23:40:35 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command
'Command'
Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command
'Command'
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Scheduled a registration timeout
for 66.153.22.16 id #17818
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 606: Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 606: Match
Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 607: Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 607: Match
Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Registration successful
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Cancelling timeout 17818
Jan 30 23:41:16 DEBUG[6245] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]'
Jan 30 23:41:30 VERBOSE[17267] logger.c: -- Attempting call on
Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 (Retry 1)
Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Executing
NoOp(Local/[EMAIL PROTECTED],2, Calling
SIP/[EMAIL PROTECTED]) in new stack
Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Executing
Dial(Local/[EMAIL PROTECTED],2, SIP/[EMAIL PROTECTED]|45|
M(say-call-2-digits^17182335097)g) in new stack
Jan 30 23:41:30 DEBUG[17269] chan_sip.c: Setting NAT on RTP to 0
Jan 30 23:41:30 DEBUG[17269] chan_sip.c: Outgoing Call for 16467508273
Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Called
[EMAIL PROTECTED]
Jan 30 23:41:30 DEBUG[6245] chan_sip.c: Acked pending invite 102
Jan 30 23:41:30 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jan 30 23:41:30 DEBUG[6245] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 103: Found
Jan 30 23:41:35 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jan 30 23:42:15 VERBOSE[17269] logger.c: -- Nobody picked up in
45000 ms
Jan 30 23:42:15 DEBUG[17269] chan_sip.c:
update_call_counter(16467508273) - decrement call limit counter
Jan 30 23:42:15 DEBUG[17269] chan_sip.c: Acked pending invite 103
Jan 30 23:42:15 DEBUG[17269] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Match
Found
Jan 30 23:42:15 DEBUG[17269] app_dial.c: Exiting with
DIALSTATUS=NOANSWER.
Jan 30 23:42:15 VERBOSE[17269] logger.c: -- Executing
NoOp(Local/[EMAIL PROTECTED],2, Done dialing from) in new
stack
Jan 30 23:42:15 DEBUG[17267] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jan 30 23:42:15 DEBUG[17267] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2007-01-30
23:42:15','16467508273','16467508273','callTo','ext-jjp-out', 'Local/[EMAIL
PROTECTED],1','','Dial','Local/[EMAIL
PROTECTED]/n',0,0,'FAILED',3,'','1170196890.32')
Jan 30 23:42:15 DEBUG[17267] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jan 30 23:42:15 DEBUG[17267] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2007-01-30 23:42:15','','','s','default',
'**Unknown**','','','',0,0,'FAILED',3,'','1170196935.35')
Jan 30 23:42:15 NOTICE[17267] pbx_spool.c: Call failed to go through,
reason 0
Jan 30 23:42:15 DEBUG[17269] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jan 30 23:42:15 DEBUG[17269] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2007-01-30
23:41:30','16467508273','16467508273','callFrom','ext-jjp-out', 'Local/[EMAIL
PROTECTED],2','SIP/tu3961-08196340','NoOp','Done dialing from',45,0,'NO
ANSWER',3,'','1170196890.33')
Jan 30 23:42:15 DEBUG[6245] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Match
Not
