[Asterisk-Users] SIP URI
Hello, I try to append a URI to the SIP dial syntax, however the URI were not shown in the sip debug message. I have read one of the post in the list which actualy show the URI string in the debug message (at the To: field). Is there any setting I need to set or turn on during compilation of asterisk? I have the head version of asterisk and my extension.conf setting is proveded below: exten = 777,1,Answer exten = 777,2,SetVar(VXML_URL=play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml) exten = 777,3,Dial(SIP/[EMAIL PROTECTED],10,t) exten = 777,4,Hangup SIP Debug message: *CLI dial 777 -- Executing Answer(OSS/dsp, ) in new stack Console call has been answered -- Executing SetVar(OSS/dsp, VXML_URL=play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml) in new stack -- Executing Dial(OSS/dsp, SIP/[EMAIL PROTECTED]|10|t) in new stack We're at 192.168.1.74 port 18952 Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.74:5060;branch=z9hG4bK280927bb From: asterisk sip:[EMAIL PROTECTED];tag=as2e2564e0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 07 Mar 2005 16:21:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 263 Thanks CFC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P module woes
the TE405 and TE410. They apparently can get locked up, and only a power cycle will clear it. My feeling (unsupported) is that the powercycle does a better job of forcing the far end of an E1 (e.g. the PTT's equipment) to start afresh than just reinitializing the cards. I have performed the following sequence: - Unplug the PRI. The green light stays ON! - modprobe -r wcte11xp. Green light stays ON - modprobe wcte11xp. Green light stays ON, ztcfg returns same error. - power down. At last, green light off! - power up. Red light blinking at about 1 Hz. - re-plug the PRI. Green light on immediately. - Asterisk won't start, known problem (ownership of /dev/zap should be asterisk... Another thing for the owner of Makefile). Corrected; on console, wcte1xxp says it sets/clears yellow alarm as Asterisk stops/starts. So... The driver works in any kernel I have tried, but if the card gets stuck, the driver won't take it out of that state. A power cycle, not even a reset, is required to recover functionality. Now asking at the Digium gurus, is this a software-correctable issue, and if it is, when will it get corrected? I don't think you can go to a customer and tell him to reboot the PBX if it doesn't work, like a windows 98, especially after shelling out a nice amount for the card. If anybody is interested in further testing or data, please mail me directly. Cheers, -- Alfredo Sola ASP5-RIPE ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID Functionality with POTS and Digium TDM04B
Hello, I'm interested in implementing DID functionality with the Digium TDM04B adapter. Is DID supported with POTS? Are there any caveeats or drawbacks that I should be aware before proceeding? This pbx will be implemented in eastern europe. Thanks in advance. -- Martin Spasov [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth phone as SIP handset?
On Fri, Mar 04, 2005 at 18:25:53 -0600, Jay Milk wrote: In a word - No. Generally, BT-capable phones can only control a headset or handsfree-set, but not be turned into a headset themselves. It's akin to expecting to watch TV on your remote, as it controls the TV so nicely :) There is, however, an effort to have asterisk become the headset to a BT capable phone, which would allow the phone to be used as FXO through a $5 USB/BT dongle without further hardware. Could the BT phone be used to dial numbers? What I have in mind in this. Asterisk on a PC. A BT headset connected to Asterisk. This is the audio input/output device. A BT phone connected to Asterisk too. You only use the BT phone to dial a number (send a number over BT to asterisk). So it is similar to using a phone/headset combination for mobile (GSM) communication, but now you are using internet calls instead. rvdp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Custom Development
Hey guys, Im looking for a programming or Development Team/Company to do some custom coding for Asterisk. What we need is not exactly simple. In fact, Im not sure the extent of the coding as far as technical terms go at all. Currently we have a call center with 4 phones. There will be a total of 8 people using the phones. Obviously, no more than 4 people will use them at a single time. But each phone will be used by 2 people. Just different times or different days. This sort of leads to a problem. We dont want users to be allowed to access each others extensions. This requires some coding. We currently use Polycom SoundPoint IP 600 SIP phones. We want a web-based login interface for the phone system. Basically, someone will go to a station which has a computer and a phone. They go to asterisk.mycompany.com and are prompted for a login and password. Each phone has a name. Phone1, Phone2, Phone3, Phone4. The first time they login, they should be asked to select a default phone. Once they select their phone and are logged in, Asterisk should route all calls for that users extension to the phone they have logged in to. This will also be used with the Queue system. We also need full reporting accessible via the interface. Total number of calls per hour, per day, per week, per month. Total number of minutes per call, per day, per agent, per extension, etc. I think you understand the reporting part now. Now I believe the above mentioned features are the most difficult and should at least let you gauge your ability to complete the project. If you think you can complete this in a reasonable amount of time, please do respond to this request and let me know some additional contact information. Phone contact is required if youd like the position. Thank you very much for your time and I look forward to your responses. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Exec AGI after hangup.
Hi everybody, I'm trying to implement a enhanced blacklist system using AGI and Perl,configuration in extension.conf is: exten =_numbera,1,AGI,blacklist_2_in.agiexten =_numbera,2,Answerexten =_numbera,3,AGI,xisco_1.agiexten =_numbera,4,AGI,blacklist_2_out.agi The problem that I have now, is that blacklist_2_out.agi doesn't execute. I think this is because in xisco_1.agi the call is hangup at the end. How can I do it in order to execute the AGI? Thanks in advance!!! Have a nice day. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip phone service for linux
Hello, i want to be able to use my zultys softphone to make calls pc-tp-pc and pc-to-phone, from my home, i want to install an asterisk server but at this time i need to connect to a voip service provider, can anybody tell my wich provider are the best and got good rates??? TIA Edgar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exec AGI after hangup.
please use deadagi, and try to do everything inside one agi. - Original Message - From: Dpto. Técnico (Softec) . To: asterisk-users@lists.digium.com Sent: Monday, March 07, 2005 2:08 AM Subject: [Asterisk-Users] Exec AGI after hangup. Hi everybody, I'm trying to implement a enhanced blacklist system using AGI and Perl,configuration in extension.conf is: exten =_numbera,1,AGI,blacklist_2_in.agiexten =_numbera,2,Answerexten =_numbera,3,AGI,xisco_1.agiexten =_numbera,4,AGI,blacklist_2_out.agi The problem that I have now, is that blacklist_2_out.agi doesn't execute. I think this is because in xisco_1.agi the call is hangup at the end. How can I do it in order to execute the AGI? Thanks in advance!!! Have a nice day. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Custom Development
Ken, This is exactly the sort of work we do, and we can fulfil all of these requirements. I'll drop you an email off list with more details of what we can provide. Feel free to email me at this address or phone me on +44 (0)7870 699 479. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Ken Sandell wrote: Hey guys, Im looking for a programming or Development Team/Company to do some custom coding for Asterisk. What we need is not exactly simple. In fact, Im not sure the extent of the coding as far as technical terms go at all. Currently we have a call center with 4 phones. There will be a total of 8 people using the phones. Obviously, no more than 4 people will use them at a single time. But each phone will be used by 2 people. Just different times or different days. This sort of leads to a problem. We dont want users to be allowed to access each others extensions. This requires some coding. We currently use Polycom SoundPoint IP 600 SIP phones. We want a web-based login interface for the phone system. Basically, someone will go to a station which has a computer and a phone. They go to asterisk.mycompany.com and are prompted for a login and password. Each phone has a name. Phone1, Phone2, Phone3, Phone4. The first time they login, they should be asked to select a default phone. Once they select their phone and are logged in, Asterisk should route all calls for that users extension to the phone they have logged in to. This will also be used with the Queue system. We also need full reporting accessible via the interface. Total number of calls per hour, per day, per week, per month. Total number of minutes per call, per day, per agent, per extension, etc. I think you understand the reporting part now. Now I believe the above mentioned features are the most difficult and should at least let you gauge your ability to complete the project. If you think you can complete this in a reasonable amount of time, please do respond to this request and let me know some additional contact information. Phone contact is required if youd like the position. Thank you very much for your time and I look forward to your responses. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT : Asterisk-Stat V2.0 - CDR Analyser
Hi Greg, How many calls do you have by hours ? BR, Areski On Fri, 2005-03-04 at 23:59, Cirelle Internet Products wrote: Areski wrote: Dear ALL, As everybody seems to like very much Asterisk-Stat, I decided to make couples of improvements... so here we go with a new version :D FEATURES : - CDR report (monthly or daily) - monthly traffic reports (pie graph) - DAILY LOAD !!! - compare call load with previous days - many criterias to define the report - export CDR report to PDF - export CDR report to CSV - support MYSQL POSTGRESQL - etc... Better to check out the screenshot: http://areski.net/asterisk-stat-v2/about.php Waiting for your feedbacks! Enjoy and have a good weekend, Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belad Arezqui Web: http://areski.net/ Email: areski ($alt) gmail ($dot) com -_-_-_-_-_-_-_-_-_-_-_-_-_-_ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users it appears the image created by graph_hourlydetail.php cannot be displayed because it has errors it works fine if there are no calls for the hour chosen. also, you might consider modifying the querys to include multiple categories, for example a good query would be one that displays the calls made to a particular destination and calls received from a particular source, not necessarily the same number. (example, I call an 800 number to report a problem and open a tickey, all calls returned to me are from a totally different number). The graph thingy, I have no idea why it contains errors regards greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P module woes
On 7 Mar 2005, at 08:42, Alfredo Sola wrote: I don't think you can go to a customer and tell him to reboot the PBX if it doesn't work, like a windows 98, especially after shelling out a nice amount for the card. No, absolutely not. My powercycle advice is only relevant when you change the config. Once you (as the installer) have set up a working config, I wouldn't expect to switch the system off until you needed to physically upgrade it. Personally I think that it isn't wholly unreasonable to have to power cycle a card when you switch from T1 to E1 or even change clock sources. Others disagree, but I guess I grew up in the age when cards were covered in DIP switches and any change required a powercycle. If your card is locking up then call Digium support, you probably have a hardware problem. Either with the card or with the server. Tim. http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exec AGI after hangup.
I have try to do everything inside one agi, and works fine. But I would like to know if it's possible to do it like seems (or an aproach)in the extensions.conf. Tnks a lot Bashir. - Original Message - From: Bashir Ullah - www.Lamsre.Com To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, March 07, 2005 11:26 AM Subject: Re: [Asterisk-Users] Exec AGI after hangup. please use deadagi, and try to do everything inside one agi. - Original Message - From: Dpto. Técnico (Softec) . To: asterisk-users@lists.digium.com Sent: Monday, March 07, 2005 2:08 AM Subject: [Asterisk-Users] Exec AGI after hangup. Hi everybody, I'm trying to implement a enhanced blacklist system using AGI and Perl,configuration in extension.conf is: exten =_numbera,1,AGI,blacklist_2_in.agiexten =_numbera,2,Answerexten =_numbera,3,AGI,xisco_1.agiexten =_numbera,4,AGI,blacklist_2_out.agi The problem that I have now, is that blacklist_2_out.agi doesn't execute. I think this is because in xisco_1.agi the call is hangup at the end. How can I do it in order to execute the AGI? Thanks in advance!!! Have a nice day. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer questions
Dear all I am trying to work out how make call trasfer work the way I want is I am the called party I want to transfer a call so I press # and enter the ext but then it disconnects me this is a blind transfer how do I make it so its not a blind transfer so i can talk to the person before i transfer the call...and go backl to the orig caller if the transfered to ext doesnt answer also can the caller hear MOH while I am talking to person I am transfering the call to what would I need to do this just point me in the right direction and i'll go read some more... I using so far is T in dial() Thanks sorry for the noob question Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BroadVoice configuration changes for Outbound
Hi All,
My version of asterisk is Asterisk CVS-HEAD-03/07/05-17:14:42
I get this error with broadvoice.
-- Executing Dial(SIP/10.217.84.12-0816c7d0,
SIP/broadvoice/011612464823xx) in new stack
Mar 7 18:52:44 NOTICE[794]: app_dial.c:936 dial_exec_full: Unable to create
channel of type 'SIP' (cause 3)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/10.217.84.12-0816c7d0, ) in new stack
== Spawn extension (default, 2011612464823xx, 2) exited non-zero on
'SIP/10.217.84.12-0816c7d0'
owl*CLI exit
does anyone know what I am doing wrong?
thanks
Dinesh.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang S.
Rupprecht
Sent: Monday, March 07, 2005 2:26 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
[EMAIL PROTECTED] (Dan Weber) writes:
On Sat, 5 Mar 2005, Wolfgang S. Rupprecht wrote:
Does broadvoice participate in e164.{arpa,org,info}?
Yes
Does this change mean that non-customers can't call broadvoice
customers with a pure SIP call by routing the call to
sip.broadvoice.com?
Calls can be made to broadvoice phones by phonenumber@sip.broadvoice.com
(From a security standpoint what is the difference between calling the
BV customer directly vs over the TELCO lines? Perhaps I'm missing
something, but better/cheaper/faster to cut out the telco middleman.)
Much cheaper over internet vs. telco.
That's great news! I had a sinking feeling when I heard the words
authenticated invite.
Unfortunately some large voip companies (cough cisco) are locking down
their sip servers to only talk to established peers. Perhaps I'm
missing something crucial, but these companies still have DID numbers
for their employees, so locking down the sip server just forces the
call to go out via the PSTN.
So are BV customers listed in the in e164.org dns zone (or some other
publicly accessible routing database)? I would love to have some way
to bypass the telco when calling friends without having to put a
by-hand entry into asterisk for each person that can accept direct
calls via some voip proxy.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
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Re: [Asterisk-Users] voicemail volume
On Asterisk 1.0 with a 4-port Digium FXO card, voicemails from the PSTN have volume so low they often can't be heard. Worse, callers sometimes get cut off in the middle of leaving a message. It is extremely frustrating to hear ...and my number is...END OF MESSAGE A search of the archives shows this is known bug: http://bugs.digium.com/bug_view_page.php?bug_id=0002023. I'm relatively new to * and don't know what parameters I can tweak to fix this. For example, where does pstnVMgain=5 go? And are there other parameters I can use to fix this problem? The problem has never been addressed; there are no parameters to fix the problem either. Best you can do is to add comments to bug 2023. The pstnVMgain parameter was a suggestion, but its never been implemented. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Fritz Capi isdn PBX integration : Can I dial out on any MSN I declare ?
Hi, I'm integrating Asterisk to legacy PBX via ISDN router. If I want to call legacy PBX internal extension I need to specify MSN as caller id and local number to call. I wonder if I can cal out via Fritz CAPI on any msn I want, or are there any limitations - I've read something about 5 MSNs limitation and wonder if it still holds ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoIP Problems?
On Mon, Mar 07, 2005 at 12:10:48AM +, Mike Dent wrote: BT providing IAX2 and SIP termination? Hmmm, maybe one day. Telstra (BTs equiv in Australia) is trialling a VoIP service. Unfortunatly, it's not quite clear what services they'll be providing... -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoIP Problems?
Asterisk guy wrote: www.mutualphone.com This company only accepts CC via PayPal doesn't sound good to me, right up there with shopping on ebay. No published address, service by calling card. Not sure about this one. Lots of red flags. I guess if it sounds too good to be true, it is. My 2cents Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: What my IAXy could have been...
Quoted message 11:26 2005.03.03 -0500, from Time Bandit: Never bought from them, never played with the stuff, but check them out anyway : http://www.iaxtalk.com/ Quoted message 11:21 2005.03.03 -0600, from Nik Martin: http://www.gumstix.com There's a grass roots IAX based phone starting up using these awesome Linux boxes. BOA web server, IAXcomm, speech recognition, bluetooth headset, etc. Really nice, and a chance to build the IAXy you always wanted. Quoted message 18:38 2005.03.03 +, from C. Tomlinson: I found http://sourceforge.net/mailarchive/forum.php?thread_id=6720059forum_id=38940 which was the most informative. Only a couple of mention on this list. Thanks for the various details, guys - will check out those links as well and see what i can dig up... I had previously also found these URLs: http://www.farfon.com/ http://ipphone.eezeephone.com/ Looks like all URLs on IAX-capable phones, http://www.iaxtalk.com/ included, point to China. Interesting... Regards: Hendrik -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer questions
At 05:44 AM 3/7/2005, you wrote: Dear all I am trying to work out how make call trasfer work the way I want is I am the called party I want to transfer a call so I press # and enter the ext but then it disconnects me this is a blind transfer how do I make it so its not a blind transfer so i can talk to the person before i transfer the call...and go backl to the orig caller if the transfered to ext doesnt answer also can the caller hear MOH while I am talking to person I am transfering the call to what would I need to do this just point me in the right direction and i'll go read some more... I using so far is T in dial() Thanks sorry for the noob question also tried the following without luck [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 it still seems to want to accept only # as transfer I am running Asterisk CVS-v1-0-03/07/05-06:50:06 Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help needed
Hello all, I Have to install an asterisk based PBX on a large Bussines, about 200 extensions, where the phone is a very critical service, this bussines need to be called and call the whole day. I am thinking to install two asterisk servers with the same config, and if one of them will be broken the otrer one takes the control of all the calls. Actualy, I do not know how would be the best way to do that, via hardware (buying a especific machine)(witch one), via software (for example rsinc, or witch software soulh I use), or other vay. What do you think about that? Witch way do you prefer? How do you do that? Any clue will be wellcomed. Ismael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoIP Problems?
Jay Milk wrote: snip and you need rock-solid performance, there are a couple of contenders out there. My most problem-free provider so far has been Vonage -- they're not very flexible, and not very open to work with their customers, but that's probably why their service has the best uptime of all the ones I used so far. Broadvoice -- read thread. Iax.cc started off promising, but it's getting spotty in places. Myphonecompany.com so far (going on three weeks) has a solid track record. Only one issue so far, and that was on my end. Aren't these just Retailers? Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel.conf and multiple T1 woes
I eventually was able to straighen out this mess by recompiling the zaptel drivers. Of course when I did that, groups broke (for some reason.) So I did a full reinstall and now it all works fine. Ken Godee wrote: I could be wrong but. Wouldn't the channel numbering follow more along these lines? That's assuming you said that you've got the first span up which would mean the TE405P is card 1, otherwise it could be card 2. It would follow that scheme if we had full T's for voice. Someone decided to get two T1's and split them 50/50 with voice and data, rather than separate voice and data T's. Hence why I am using channels 12-23 for voice. Also, what do you mean by I inherited them ? Where did they come from? Are you moving them from another piece of equipment? If so, are you sure the second span even has a D channel? Maybe it was part of an NFAS group? I am new to the company that rolled this out. So essentially I had their entire Asterisk project dropped in my lap, including the wacky T1 setup. I am sure the second T has a D channel since it is notated on our paperwork, and it did not sync up correctly on our previous install. As I said, a flatten and reinstall fixed all of the problems, so I am not quite sure what happened. But it at least works now. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P module woes
On March 7, 2005 03:42 am, Alfredo Sola wrote: - Unplug the PRI. The green light stays ON! - modprobe -r wcte11xp. Green light stays ON The card is *most certainly* locked up. - power up. Red light blinking at about 1 Hz. - re-plug the PRI. Green light on immediately. This is mostly normal. Sometimes it will go green immediately, sometimes it will go orange for a few seconds, then green. It all depends on the exact state of the card/driver. - Asterisk won't start, known problem (ownership of /dev/zap should be asterisk... Another thing for the owner of Makefile). Corrected; on Not true; this is only a known issue for those who do not run asterisk as root. console, wcte1xxp says it sets/clears yellow alarm as Asterisk stops/starts. What happens when you plug a loopback plug in to the T1? (pin 1-4, 2-5) -- the TE110P should go up and stay up, and if it's a PRI it should be complaining about CPE/CO side. So... The driver works in any kernel I have tried, but if the card gets stuck, the driver won't take it out of that state. A power cycle, not even a reset, is required to recover functionality. Call Digium and either get the card RMA'd or let them get in to your machine ot see what they can do. (likely the latter before the former). This is *not* normal. I don't think you can go to a customer and tell him to reboot the PBX if it doesn't work, like a windows 98, especially after shelling out a nice amount for the card. Agreed 100%. This is either a mainboard/card compatiblity issue or a dead card. Either way, Digium can help you if you call/email their support and get the procedure started. Support is included in that $500 price tag. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENT : Asterisk-Stat V2.0 - CDR Analyser
Areski wrote: Hi Greg, How many calls do you have by hours ? BR, Areski Hi Areski, Some hours 0 most hours 1-3 with bursts of 7 - 14 Not a lot of traffic on the dates I had checked. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help needed
On 7 Mar 2005, at 12:10, [EMAIL PROTECTED] wrote: Hello all, I Have to install an asterisk based PBX on a large Bussines, about 200 extensions, where the phone is a very critical service, this bussines need to be called and call the whole day. I am thinking to install two asterisk servers with the same config, and if one of them will be broken the otrer one takes the control of all the calls. Actualy, I do not know how would be the best way to do that, via hardware (buying a especific machine)(witch one), via software (for example rsinc, or witch software soulh I use), or other vay. What do you think about that? Witch way do you prefer? How do you do that? Any clue will be wellcomed. Ah, It depends on a few factors. 1) What is an acceptable downtime ? 2) What is an acceptable frequency of downtime ? 3) What technical resources are available on/off site? 4) How much is it worth ? Others will no doubt chime in with higher-end solutions, but consider this as a basic starting point: Lets assume that acceptable downtime is Max 4 hours and acceptable frequency is once every 2 years. Also let's say there are competent PC staff on site, but no Linux skills. Finally imagine that the customer says that it's worth $5k/pa to hit those targets. The easiest way to do this is to buy quality hardware (MTBF of 5 years say), create a good backup scheme and train the onsite folks how/when to do a restore. Then either buy a duplicate system and put it in a cupboard or sign a 2 hour maintenance contract with a hardware vendor. This way, when they have a problem, they wheel out the spare, load the backup, swap the cables and away you go. The _huge_ advantage of this is simplicity. I know a high availability site where 2 out of the 3 failures in 5 years were directly related to misconfigured HA :-) My favorite was when the license for the 'robust' filesystem expired ! Tim. Ismael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dock-n-talk connection to asterisk
Hi Peter. Look in last weeks (1/3/05) Sydney Morning Herald Tuesday IT liftout. They talk there about GSM gateways. It was made by Ericson I think, for around $1000. It's not meant for computer, rather as a FXO/FXS gateway to plug your house phone in for exactly the purpose you are talking about. Of course, if it is a FXO gateway, I'm sure a RJ cable (possibly crossover) will plug it in to a TD400 Digium card nicely to get what you want. I'm interested to know your progress, I have a few clients also interested in Sydney. Cheers Mike - Original Message - From: Peter Illmayer [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 05, 2005 2:06 PM Subject: [Asterisk-Users] Dock-n-talk connection to asterisk Hi ALL I'm looking for feedback on how well this unit integrates into asterisk via an ata. Is the audio quality any good as thats the first thing to upset the wife if its no good. I'm looking for a reasonably priced GSM gateway 1800mhz for use in Australia that works with an ata. Quite happy to import something that works well... Currently PSTN to mobile is $0.40c per minute and going to a selected provider, it will only cost $0.05c per minute so the savings are enormous for me, hence my interest in the DOck-n-Talk Any feedback would be very much appreciated ! -- Open WebMail Project (http://openwebmail.org) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.2 - Release Date: 4/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help needed
Ismael, I'm not going to give you a full answer, because this is a big topic, and I sell high availability systems to my own customers. Having said that, here are some ideas. This list is not definitive, and I'm sure other people will have other suggestions. You have 2 issues: 1. Keeping the configuration and data in sync. You can keep the config files in sync using rsync; that's the easy part. If you use the Asterisk internal database for persistent data, you'll need to keep that in sync. Offhand, I don't know how to do this, as I tend not to use the Asterisk DB for large systems. If you were paying me, I'd go and figure it out, but as you're not, I'm too lazy! Go search on google. If you use an SQL database, you could connect both machines to one database server, though this then makes this machine a single point of failure. You could use MySQL's master-slave replication. You could use an O/S level failover product like GFS. You could handle it at application level, writing to more than one database server (though this is really hard to get right). 2. Routing calls to the working machine. The easy and cheap way is to do the failover is by hand. Keep a warm spare running with it's configuration synchronised, then if the master fails, unplug it, and plug in the spare. If you want it automatic: If you have E1s or T1s, most PBXs and providers can detect trunks in a trunk group with loss of signal, and take that trunk out of the group. Thus you connect trunks to each machine, and when one fails, no calls are delivered to it. For VoIP, if your clients and VoIP peers can handle DNS SRV records properly, they can fail over when a machine goes down. Be warned that many products don't work properly. For other Asterisk systems, use the 'qualify' option in sip.conf so it knows when other systems are available, then different priorities in extensions.conf to fail over. If neither of the above are possible, consider using IP takeover. This is a tricky thing to make work 100% - you may need expert help. For FXS ports for analogue phones, there aren't many options, as they're dumb devices. You may be able to find some hardware box that has N FXS ports and 2N FXO ports that can route calls from analogue phones to the correct machine. Good luck! You've chosen a difficult area to work in. If you run into problems, we can offer help on a commercial basis. I've installed some very large high availability systems - voicemail clusters each with 400,000 users, 96 T1s, and no single point of failure, for instance. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ [EMAIL PROTECTED] wrote: Hello all, I Have to install an asterisk based PBX on a large Bussines, about 200 extensions, where the phone is a very critical service, this bussines need to be called and call the whole day. I am thinking to install two asterisk servers with the same config, and if one of them will be broken the otrer one takes the control of all the calls. Actualy, I do not know how would be the best way to do that, via hardware (buying a especific machine)(witch one), via software (for example rsinc, or witch software soulh I use), or other vay. What do you think about that? Witch way do you prefer? How do you do that? Any clue will be wellcomed. Ismael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer questions
also tried the following without luck [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 it still seems to want to accept only # as transfer I am running Asterisk CVS-v1-0-03/07/05-06:50:06 You are running V1.0.x stable of asterisk. Tthe attended transfer feature is only available in CVS-HEAD, which at some point (June ?) will become 1.1.x stable ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoIP Problems?
on 3/4/05 22:18, Tim at [EMAIL PROTECTED] wrote: Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider? I wonder if LiveVOIP uses Voice Conduits or vice-versa. Voice Conduits has been down since Friday too. :-( -- R.J. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS compile error utils.c
Hi.. I get the following error when compiling the lastest CVS utils.c:405: undefined reference to `__use_ast_pthread_create_instead__' due to the fact I dont know c I thought what the heck and took a look at line 405 return pthread_create(thread, attr, start_routine, data); and changed it to ast_pthread_create(thread, attr, start_routine, data); and it compiledbut when running asterisk it gives me a nasty bus error and dies :/ now i'm feeling stupoidcan someone help??? :) Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP and Astersik
I have FXO (DIGIUM) with Asterisk (PBX). How can I use SNMP in Asterisk to access FXO? I need to known if FXO has the LINE with PSTN free to new phone call. Is this possible? How? There is no support/code for snmp in asterisk. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Open files / socket leak
Title: Open files / socket leak We're using STABLE CVS-Nv1-0-5-02/24/05 and we've been noticing that sometimes there's a socket leak on REGISTER SIP messages. We've seen it happen only on customers using Sipura SPA2100 ATAs. If I issue a sip show channels, I see thousands of zombie channels. If I look into the details, that's what I get - actually one single sip show channel channelID returns thousands of these: * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED]: 520 REGISTER Our Codec Capability: 12 Non-Codec Capability: 1 Their Codec Capability: 0 Joint Codec Capability: 0 Format unknown Theoretical Address: x.x.x.x:5060 Received Address: x.x.x.x:5060 NAT Support: RFC3581 Our Tag: 715659627 Their Tag: SIP User agent: Need Destroy: 0 Last Message: Promiscuous Redir: No Route: N/A DTMF Mode: rfc2833 The sequence number (ie. 520) increases by 1 every time. After a while, I run out of files and I have to restart asterisk. I have temporarily solved the problem by issuing a ulimit -n 8192 in safe_asterisk, but that's not a solution, since I will eventually reach that limit as well. Is there a way to fix this? We're running RHEL4 and we have about 300 customers registered all the time. Thank you very much -Manuel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
I've fought this all weekend. Friday, they couldn't take an order because the credit card thing on the website was broken. Saturday, I got an account. Incoming works, put the phonenumber at the end of the register string and then place that number as an extension in your broadvoice context. Outbound still doesn't work. I've tried everything on this list and everything I could find on the wiki and all other lists. Going home. Sympathetic responses greatly appreciated. BTW, who else does flat-rate BYOD? I had BV service for a while, but moved on due to limited dsl bandwidth on my end and their support for sip only. Now using iax with teliax.com and really haven't had any problems. While I had BV service, I noticed issues with registration as well, but those seemed to be related to exactly which BV server I was trying to register with. Through experimenting I found one of their servers to be almost 100% reliable and used it for several months. Not sure if that has changed in the last month or so though. (I had to place 147.135.8.128 sip.broadvoice.com in /etc/hosts.) Judging from past user posts, there seems to be a fair number of users that have asterisk behind a firewall and apparently don't understand what is needed to properly configure * for this. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP and Astersik
How can I check how many lines PSTN I have free to do phone call? On Mon, 7 Mar 2005, Rich Adamson wrote: I have FXO (DIGIUM) with Asterisk (PBX). How can I use SNMP in Asterisk to access FXO? I need to known if FXO has the LINE with PSTN free to new phone call. Is this possible? How? There is no support/code for snmp in asterisk. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD and SIPPHONE problems after upgrading toCVSHEAD - VERIFIED
Mike Matthews wrote: This works for me both incoming and outgoing w/Sipphone. Note there is NO username, secret entries in the peer definition. I am using * vers 1.05 register=1747nnn:[EMAIL PROTECTED]/1747xxx ; note:extension in extensions.conf matches for incoming Thank you very much for weighing in, I was getting paranoid that everyone was blacklisting the few posts I make a year :-) I too can get it to work on Stable (versions 1.0.3 and 1.0.6), so I'm not surprised to hear your results. I also had to add an insecure=very, which is disappointing, but since it is hard-coded to a particular IP address, I guess it's not as awful as it could be otherwise. That said, I don't think I dropped the username from the CVS HEAD test, but I did add insecure=very, which still failed. So, I continue to maintain that _something_ has changed in CVS HEAD which makes incoming and outgoing calls fail to SIPPHONE, and _outgoing_ calls via IAX2 fail to FWD (for me), while incoming calls from FWD via IAX2 definitely continue to work for me. For the moment, it remains a mystery... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip not 100% RFC3665 compliant - re-REGISTERs fail.
Greetings, For the past 2 months I've been struggling with registration problems with asterisk+external FXS/FXO gateways (www.addpac.com) that use RFC3665 re-registration procedure. This problem occured for devices with more than one FXS port with a set non-empty password. Those gateway attempt to re-register after the initial register timeout period expires fully compliant with RFC3665, clause 2.2 (http://www.zvon.org/tmRFC/RFC3665/Output/chapter2.html#sub2), but asterisk fails to authenticate them. The 1st FXS port of the device always registers successfuly (although still uses same RFC3665, clause 2.2 procedure), but the remainder fail miserably. Using an account/username with an empty password for the affected ports fixes the problem - so this is something with www-digest method (?). I've spent 2 weeks debugging this with addpac development team, and the same device authenticates flawlessly with Sonus Proxy Server, SNOM Proxy Server, LongBoard Proxy Server, Nortel Proxy so this seems to be a problem with chan_sip. I'm hesitant to post the long sip debug outputs to the mailing list to conserve the bandwidth. More info and sip debugs are available at http://bugs.digium.com/bug_view_page.php?bug_id=0003726 Is there anyone else with the same problem? regards, Vahan begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardphone deployment recommendation
Thanks for your replies. My main concern is to keep the price down. If the BudgeTones are crap phones, which previous posts to this mailing list seem to indicate, and we have to replace them often, then the price for a better phone would be worth it. I don't think we need 3-way calling either, and as stated already, nothing a proper dialplan can't fix. I'm going to try getting in a couple of those Sipura 841s for testing. Thanks for that suggestion. -- Dana ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to get (cheap) VoIP
Hi, I would like to deploy a (very) small PBX at my place, so that I can stop answering phones for my kids or my wife, using distinctive ringings. I read that, using a modem,I can use a standard phone line, and convert that as input for Asterisk PBX, right? Also, where can I get VOIP phones? Does Asterisk work with *any* PBX phones? Any brands to recommand? Thanks in advance for you answers! Christian Faucher ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP howto
Hey there, I'm an asterisk newbie and have just joined this mailing list. I have to use asterisk as a call agent that supports MGCP requests. I'm reading the documentation from asteriskdocs and voip-info.org but those cover more specifically only IAX and SIP configuration. I'd really appreciate it if someone can tell me where to find more detailed documentation on how to configure asterisk to work with MGCP. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk supports VXML?
Hi all where can I find infos aboutthis VXML intepreterfor asterisk? Thanks Marco Hi Foong, That's a good question you've put out there. Yes, Asterisk supports VXML andhere's how it's done; Firstly in the SIP.conf, you need to have your VXML application/browserdefined; sip.conf: [vxmlapp] type=friend insecure=yes username=777 reinvite=no host=123.45.67.8 Then in the EXTENSIONS.conf it will look like this; extensions.conf: exten =777,1,Setvar,VXML_URL=voicexml=http%3A%2F%2F123.45.67.20%3A6969%2Fhellovxml%2Fhellovxml exten = 777,2,Dial,sip/vxmlapp|10 exten = 777,3,HangUp Hope this'll clear your thoughts. Cheers! Lilantha Karunaratne MSCSTel: (65) 90403497 _ From: asterisk-users-bounces at lists.digium.com[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chee FoongSent: Friday, February 25, 2005 10:17 AMTo: asterisk-users at lists.digium.comSubject: [Asterisk-Users] asterisk supports VXML? Hello,Does asterisk supports VXML?Couldn't find much resource on that on google and wiki.ThanksFoong ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk supports VXML?
Marco, There isn't. When asked about VoiceXML by my customers, I recommend using a Cisco router for VXML interpretation, and SIP to integrate it with Asterisk. There are a wide variety of PC based proprietary VXML browsers that you can use instead of Cisco. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Marco Parisotto wrote: Hi all where can I find infos about this VXML intepreter for asterisk? Thanks Marco Hi Foong, That's a good question you've put out there. Yes, Asterisk supports VXML and here's how it's done; Firstly in the SIP.conf, you need to have your VXML application/browser defined; sip.conf: [vxmlapp] type=friend insecure=yes username=777 reinvite=no host=123.45.67.8 Then in the EXTENSIONS.conf it will look like this; extensions.conf: exten = 777,1,Setvar,VXML_URL=voicexml=http%3A%2F%2F123.45.67.20%3A6969%2Fhellovxml% 2Fhellovxml exten = 777,2,Dial,sip/vxmlapp|10 exten = 777,3,HangUp Hope this'll clear your thoughts. Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 _ From: asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users [mailto:asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users] On Behalf Of Chee Foong Sent: Friday, February 25, 2005 10:17 AM To: asterisk-users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users Subject: [Asterisk-Users] asterisk supports VXML? Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and ISDN
Title: SIP and ISDN I have set up an Asterisk PBX server and can make calls between endpoints using both the SIP and IAX protocols. Iam using X-Lite softphone to make SIP calls and DIAX softphone to make IAX calls. The next step is to get an ISDN line connected and ISDN phone able to make calls to either a SIP or IAX softphone. So far I have managed to install an AVM Fritz card along with the drivers and CAPI. I can attempt to make a call to a softphone but the call cannot be connected. The Asterisk PBX does process the call and displays the msn that the ISDN phone is tring to call but the softphone does not ring and no call is established. Any configuration ideas on how I can get this to work? Is there anything I have missed? Here is a diagrammatical explanation: PC - Softphone | Ethernet Line Asterisk PBX | ISDN line ISDN phone Any suggestions will be a great help. Philip Lee + pp 101D Gemini Buildings Adastral Park Martlesham Heath Ipswich Suffolk IP5 3RE ( (Work): 01473 648158 ( (Mobile): 07793738044 * (Work): [EMAIL PROTECTED] * (Home): [EMAIL PROTECTED] BTexact Technologies is a trademark of British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 180 This electronic message contains information from British Telecommunications plc which may be privileged or confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the numbers or address above) immediately. Activity and use of the British Telecommunications plc E-mail system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip not 100% RFC3665 compliant - re-REGISTERsfail.
[EMAIL PROTECTED] wrote: Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip not 100% RFC3665 compliant - re-REGISTERsfail.
There are issues with Asterisk chan_sip. Have a look at bug 759 at bugs.digium.com. Comments in the feature report and source code like those below probably go a way to explain your problems. I don't know how much of this test version has been back-ported to chan_sip, however the chan_sip2.c with a November 2004 CVS seems to work quite well. Olle Johansson has suspended work on this for now due to workload and it probably won't compile any more against the latest CVS due to changes elsewhere. Peter * Added support for WWW-auth for registrations (according to SIP RFC). * + WARNING: This version changes a lot of functionality in regards *to authentication, we use the digest auth username to check *credentials for INVITES, not the username@ in the From: URI *INVITEs are authenticated this new way, not REGISTER/SUBSCRIBE *yet -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian Sent: 07 March 2005 13:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip not 100% RFC3665 compliant - re-REGISTERsfail. Greetings, For the past 2 months I've been struggling with registration problems with asterisk+external FXS/FXO gateways (www.addpac.com) that use RFC3665 re-registration procedure. This problem occured for devices with more than one FXS port with a set non-empty password. Those gateway attempt to re-register after the initial register timeout period expires fully compliant with RFC3665, clause 2.2 (http://www.zvon.org/tmRFC/RFC3665/Output/chapter2.html#sub2), but asterisk fails to authenticate them. The 1st FXS port of the device always registers successfuly (although still uses same RFC3665, clause 2.2 procedure), but the remainder fail miserably. Using an account/username with an empty password for the affected ports fixes the problem - so this is something with www-digest method (?). I've spent 2 weeks debugging this with addpac development team, and the same device authenticates flawlessly with Sonus Proxy Server, SNOM Proxy Server, LongBoard Proxy Server, Nortel Proxy so this seems to be a problem with chan_sip. I'm hesitant to post the long sip debug outputs to the mailing list to conserve the bandwidth. More info and sip debugs are available at http://bugs.digium.com/bug_view_page.php?bug_id=0003726 Is there anyone else with the same problem? regards, Vahan This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to get (cheap) VoIP
Christian faucher wrote: I read that, using a modem,I can use a standard phone line, and convert that as input for Asterisk PBX, right? Not that simple, not every modem, but yes. Also, where can I get VOIP phones? eBay ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk supports VXML?
Marco, /me goes back and reads the rest of the email as he should have in the first place. What they're talking about is an external VoiceXML browser which they connect to over SIP, just as I've mentioned with Cisco. I don't know which browser though. Time for me to get stronger glasses, I think. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Alistair Cunningham wrote: Marco, There isn't. When asked about VoiceXML by my customers, I recommend using a Cisco router for VXML interpretation, and SIP to integrate it with Asterisk. There are a wide variety of PC based proprietary VXML browsers that you can use instead of Cisco. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Marco Parisotto wrote: Hi all where can I find infos about this VXML intepreter for asterisk? Thanks Marco Hi Foong, That's a good question you've put out there. Yes, Asterisk supports VXML and here's how it's done; Firstly in the SIP.conf, you need to have your VXML application/browser defined; sip.conf: [vxmlapp] type=friend insecure=yes username=777 reinvite=no host=123.45.67.8 Then in the EXTENSIONS.conf it will look like this; extensions.conf: exten = 777,1,Setvar,VXML_URL=voicexml=http%3A%2F%2F123.45.67.20%3A6969%2Fhellovxml% 2Fhellovxml exten = 777,2,Dial,sip/vxmlapp|10 exten = 777,3,HangUp Hope this'll clear your thoughts. Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 _ From: asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users [mailto:asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users] On Behalf Of Chee Foong Sent: Friday, February 25, 2005 10:17 AM To: asterisk-users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users Subject: [Asterisk-Users] asterisk supports VXML? Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2-Ring Delay for CLID
Hello All, Need a little direction, please. I have searched the lists, WIKI, and googled a problem that I'm sure I'm overlooking. I understand why Asterisk/Zaptel waits two rings to answer (caller ID must be sent) but can I reduce the amount of time it takes before Asterisk/Zaptel answers? In other words, I'm not concerned about Caller ID and want the line answered as quickly as possible. Matthew Machen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2-Ring Delay for CLID
Yep, disable callerid in zapata.conf On Mon, 7 Mar 2005 08:56:39 -0600, Machen, Matthew T. [EMAIL PROTECTED] wrote: Hello All, Need a little direction, please. I have searched the lists, WIKI, and googled a problem that I'm sure I'm overlooking. I understand why Asterisk/Zaptel waits two rings to answer (caller ID must be sent) but can I reduce the amount of time it takes before Asterisk/Zaptel answers? In other words, I'm not concerned about Caller ID and want the line answered as quickly as possible. Matthew Machen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2-Ring Delay for CLID
Matthew, if you don't use clid then just comment out the clid references using a semi colon ; Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Machen, Matthew T. Sent: Monday, March 07, 2005 9:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 2-Ring Delay for CLID Hello All, Need a little direction, please. I have searched the lists, WIKI, and googled a problem that I'm sure I'm overlooking. I understand why Asterisk/Zaptel waits two rings to answer (caller ID must be sent) but can I reduce the amount of time it takes before Asterisk/Zaptel answers? In other words, I'm not concerned about Caller ID and want the line answered as quickly as possible. Matthew Machen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio pausing over IAX trunk
Florian Overkamp wrote: Hi Steve, -Original Message- I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to make no difference either. The pause happens every few seconds, and is regular. Not unless you can describe the problem more clearly. Which direction does this happen in, what exactly are these clients you're talking about, and what is does the network look like between the endpoints. Okay, in my scenario it's like this: SIP or MGCP phone (mixed env.) - Asterisk box - IAX - Asterisk box - PSTN or other Asterisk box We notice users complaining of the fact that the remote end (PSTN) complained about audio drops, while the local user keeps hearing everything. I am not entirely sure if it is just that direction, because I hear noticeable crackles during the call from my (user) end too. This appears to happen especially when the asterisk boxes involved have a few calls happening, when its nice and quiet on the box, things seem ok. This kind of thing is not or hardly noticable when calling yourself, which makes diagnosis difficult. I've discussed this with other people on the list, and we notice the following: IP links are _not_ congested and latency is very stable, so we are not looking at a network issue. Others have observed that changing the protocol from IAX2 to SIP is a good workaround. I have not yet been able to confirm this because we are tied to Asterisk-stable which does not yet have a very useable SIP dialling format. It's very hard to get a good handle on this issue, because it pretty much requires a multihomed production box to work with :-( I'm not sure exactly what your problem is, but I think that the new JB may help; at the very least, you could run iax2 show netstats, and get an idea of what the right-most asterisk box is seeing. Also my latest patchset would keep the JB out of the loop on the left-most asterisk box when it's bridging, and on the right-most box, it would use it if you were bridging to the PSTN (i.e. via zap, I guess), and would not use it when you were bridging to another asterisk box via a VoIP protocol.. See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532 -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and ISDN
On 7 Mar 2005, at 14:27, [EMAIL PROTECTED]> wrote: I have set up an Asterisk PBX server and can make calls between endpoints using both the SIP and IAX protocols. Iam using X-Lite softphone to make SIP calls and DIAX softphone to make IAX calls. The next step is to get an ISDN line connected and ISDN phone able to make calls to either a SIP or IAX softphone. So far I have managed to install an AVM Fritz card along with the drivers and CAPI. I can attempt to make a call to a softphone but the call cannot be connected. The Asterisk PBX does process the call and displays the msn that the ISDN phone is tring to call but the softphone does not ring and no call is established. Any configuration ideas on how I can get this to work? Is there anything I have missed? Here is a diagrammatical explanation: PC - Softphone | x-tad-smallerEthernet Line/x-tad-smallerAsterisk PBX |x-tad-smaller ISDN line/x-tad-smallerISDN phone Any suggestions will be a great help. You'll need to send us some debug logs and snippets of config files for us to help you with this problem. At a pure guess I'd say you need to add some lines to extensions.conf we will know more if you send more info. Tim. http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardphone deployment recommendation
On Mon, March 7, 2005 8:59 am, Dana Olson said: I'm going to try getting in a couple of those Sipura 841s for testing. Thanks for that suggestion. FYI, I have a site with 6 of the SPA-841 units in plave and they were working find until last week. On of them licked up and now cannot get past Initializing Network on boot. It's on it's wayback to Sipura for replacement. A more uncomfortable issue is that the speaker phones were found to be working very poorly. The speakerphone user is just about inaudable to the user on the other end of the call. This is the case with all of the units I have. I had them all running the lates firmware from the website. In Sipura's defense, they responded within about 15 minutes to my support email with a link to a test version of the firmware which improved things but didn't completely fix it. My gut feel is that Sipura is still learning with their first hard phone. The price is great for the feature set and I have no other serious complaints about the phone. I'd like to see the buttons improved and the display be tiltable for better viewing. We got used to both of these quickly though. We'll be overjoyed when the speakerphone wrinkles get ironed out. $0.02, -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3 stream for MOH
Any suggestions how can I get asterisk to play MOH (music on hold) a MP3 radio stream from the internet (http:// location) instead of a MP3 file in the mphmp3 folder? I tried puttingdefault = quietmp3:http://www.waixwave.com/pacnet.pls instead of default = quietmp3:/var/lib/asterisk/mohmp3 but did not work got message NOTICE[25564]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Any suggestions how to get the mp3 stream work? Thanks. CJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bluetooth phone as SIP handset?
Not as such, no. You may be able to take a java-enabled phone and write a remote-control app for it, which will allow you to do what you want. I hardly doubt it's worth the effort though -- then you're already in for the BT handset, etc. If you want cordless asterisk, get yourself a cheap cordless phone and an ATA. -Original Message- From: Ronald van der Pol [mailto:[EMAIL PROTECTED] Sent: Monday, March 07, 2005 3:37 AM To: Jay Milk Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] Bluetooth phone as SIP handset? On Fri, Mar 04, 2005 at 18:25:53 -0600, Jay Milk wrote: Could the BT phone be used to dial numbers? What I have in mind in this. Asterisk on a PC. A BT headset connected to Asterisk. This is the audio input/output device. A BT phone connected to Asterisk too. You only use the BT phone to dial a number (send a number over BT to asterisk). So it is similar to using a phone/headset combination for mobile (GSM) communication, but now you are using internet calls instead. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup detection failing
If your problem is the same as mine then you need to use busydetect. In Turkey, we don't have polarity reversal, and signalling tones are quite different. But just enabling busydetect in zaptel.conf did not help me, it may work for you though. I had to change relevant compile options and some settings in dsp.c file and recompile. (Thegoal is to detect congestion tone here.) I haven't noticed any problems so far,but I am still working on it to use Martin's algorithm instead. Hope this helps, Soner - Original Message - From: Cameron Beattie To: asterisk-users@lists.digium.com Sent: Monday, March 07, 2005 1:31 AM Subject: [Asterisk-Users] FXO module in TDM400P (UK,BT) - Hangup detection failing I am based in New Zealand and am experiencing the same problem as referred to in the post "FXO module in TDM400P (UK, BT) - Hangup detection failing" from 2 November 2004 i.e. Zap/4 (being the FXO module) not detecting hangup on the PSTN line if the call is not answered on a PABX extension. Has anyone managed to find a resolution to the problem? For information: Digium TDM400P with FXS on Zap 1 2 and FXO on Zap 3 4. CVS-v1-0-01/24/05 Using fxs_ks signalling Regards Cameron ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP and ISDN
Here are some config files: sip.conf [general] register = 222:[EMAIL PROTECTED]:5060/222 register = 111:[EMAIL PROTECTED]:5060/111 port = 5060 tos=lowdelay jitterbuffer=yes maxjitterbuffer=yes maxjitterbuffer=500 maxexcessbuffer=100 bindaddr = 0.0.0.0 allow=all ;allow=ilbc ;allow=alaw context = fullaccess [111] type=friend ;host=xx.xx.xx.xx host=dynamic username=111 secret=mysecret context=sip-access1 callerid=Philip 2 111 ;reinvite=no ;caninvite=no ;qualify=500 nat=yes allow=all ;allow=gsm ;dtmfmode=rfc2833 [222] type=friend ;host=xx.xx.xx.xx host=dynamic username=222 secret=mysecret context=sip-access2 callerid=Philip Lee 222 ;reinvite=no ;caninvite=no ;qualify=500 nat=yes allow=all ;allow=gsm ;dtmfmode=rfc2833 capi.conf [general] ;mode=immediate ;isdnmode=multipoint nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=0 incomingmsn=0 ;overlapdial=yes ;outgoingmsn=01912500900 controller=1 ;softdtmf=1 ;accountcode= context=capi-access1 mode=immediate isdnmode=ptp devices=2 ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect= msn=9 incomingmsn=* controller=1 context=capi-access2 mode=immediate isdnmode=ptp devices=2 msn=0 incomingmsn=0 controller=1 context=capi-access1 mode=immediate isdnmode=ptp devices=2 msn=9 incomingmsn=9 controller=1 context=capi-access2 mode=immediate isdnmode=ptp devices=2 msn=0 incomingmsn=* controller=1 context=capi-access1 mode=immediate isdnmode=ptp devices=2 extensions.conf [general] static=yes writeprotect=yes [noaccess] exten = _.,1,Congestion [sip-access1] exten = 222,1,Dial(SIP/222,20,tr) exten = 1,1,Dial(IAX2/user1,20,tr) exten = 2,1,Dial(IAX2/user2,20,tr) exten = 333,1,Dial(CAPI/01912500900,30) exten = 444,1,Dial(CAPI/01912500909,30) [sip-access2] exten = 111,1,Dial(SIP/111,20,tr) exten = 2,1,Dial(IAX2/user2,20,tr) exten = 1,1,Dial(IAX2/user1,20,tr) exten = 333,1,Dial(CAPI/@01912500900,30) exten = 444,1,Dial(CAPI/@01912500909,30) [iax-access1] exten = 111,1,Dial(SIP/111,20,tr) exten = 222,1,Dial(SIP/222,20,tr) exten = 2,1,Dial(IAX2/user2,20,tr) ;exten = 1,3,Voicemail(u1) exten = 333,1,Dial(CAPI/01912500900,30) exten = 444,1,Dial(CAPI/01912500909,30) [iax-access2] exten = 1,1,Dial(IAX2/user1,20,tr) exten = 111,1,Dial(SIP/111,20,tr) exten = 222,1,Dial(SIP/222,20,tr) exten = 333,1,Dial(CAPI/@01912500900,30) exten = 444,1,Dial(CAPI/@01912500909,30) [capi-access1] exten = 1,1,Dial(IAX2/user1,20,tr) exten = 111,1,Dial(SIP/111,20,tr) exten = 2,1,Dial(IAX2/user2,20,tr) exten = 222,1,Dial(SIP/222,20,tr) exten = 444,1,Dial(CAPI/01912500909,30) [capi-access2] exten = 1,1,Dial(IAX2/user1,20,tr) exten = 111,1,Dial(SIP/111,20,tr) exten = 2,1,Dial(IAX2/user2,20,tr) exten = 222,1,Dial(SIP/222,20,tr) exten = 333,1,Dial(CAPI/01912500900,30) Also here is some debug info. This is from when I make a call to one of the msn numbers from an ISDN phone. The ISDN phone rings but the other endpoint (softphone) does not ring and no connection is established. *CLI capi debug CAPI Debugging Enabled *CLI -- CONNECT_IND ID=001 #0x0004 LEN=0028 Controller/PLCI/NCCI= 0x101 CIPValue= 0x1 CalledPartyNumber = 810 CallingPartyNumber = default CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default Mar 7 15:43:29 NOTICE[10058]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=001 #0x0004 LEN=0028 Controller/PLCI/NCCI= 0x101 CIPValue= 0x1 CalledPartyNumber = 810 CallingPartyNumber = default CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default == CONNECT_IND (PLCI=0x101,DID=0,CID=(null),CIP=0x1,CONTROLLER=0x1) -- INFO_IND ID=001 #0x0005 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = 810 -- INFO_IND ID=001 #0x0006 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 8a -- DISCONNECT_IND ID=001 #0x0007 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x0 == DISCONNECT_IND PLCI=0x101 REASON=0 Mar 7 15:43:39 WARNING[10058]: chan_capi.c:1380 pipe_msg: unable to hangup channel on DID. Channel is NULL. -Original Message- From: [EMAIL PROTECTED] on behalf of tim panton Sent: Mon 3/7/2005 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [Asterisk-Users] SIP and ISDN
RE: [Asterisk-Users] Hardphone deployment recommendation
Psul I bought one a few weeks ago. I had the same issue with 'initialising network.' I had a very fast response back from their support desk. Solution is to unplug the network cable reset back to factory defaults from the keypad menu. I have concerns over the Sipura 841's TX audio quality - it seems to have some AGC or noise suppression which does not work at all well. If you talk quietly the phone seems to mute all TX audio. Shout and it starts sending. I have disabled silence supression. It needs 'de-Americanising' for progress tones, languages, date and time formats and also needs the SUBSCRIBE/NOTIFY support or similar for multiple line appearances. I think with a bit more work it will be a great little phone. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Dugas Sent: 07 March 2005 15:26 To: Dana Olson; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hardphone deployment recommendation On Mon, March 7, 2005 8:59 am, Dana Olson said: I'm going to try getting in a couple of those Sipura 841s for testing. Thanks for that suggestion. FYI, I have a site with 6 of the SPA-841 units in plave and they were working find until last week. On of them licked up and now cannot get past Initializing Network on boot. It's on it's wayback to Sipura for replacement. A more uncomfortable issue is that the speaker phones were found to be working very poorly. The speakerphone user is just about inaudable to the user on the other end of the call. This is the case with all of the units I have. I had them all running the lates firmware from the website. In Sipura's defense, they responded within about 15 minutes to my support email with a link to a test version of the firmware which improved things but didn't completely fix it. My gut feel is that Sipura is still learning with their first hard phone. The price is great for the feature set and I have no other serious complaints about the phone. I'd like to see the buttons improved and the display be tiltable for better viewing. We got used to both of these quickly though. We'll be overjoyed when the speakerphone wrinkles get ironed out. $0.02, -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SJphone on PDA registering with Asterisk???
Well, given your setup and the fact that you aren't seeing anything on the console with verbose debugging on, I'm going to guess there is a network/routing issue here. I'd try getting the PDA on line and just doing some simple ping tests to the 192.168.250.x network from it. (including to the * server). If you can reach it then is surely should let you register or at least give you info on the console. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Sunday, March 06, 2005 8:59 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SJphone on PDA registering with Asterisk??? James Pooton wrote: I'm all so using SJphone on my x50v, works surprisingly well :). Is voip.elmit.com also in the 192.168.1.X NAT space that your PDA is in? There might be the problem: I have the server at two ethernet cards reachable: Extern with a public IP Intern with 192.168.250.20 on this internal LAN is a wireless accesspoint, which in return changes the IP address to a network 192.168.1.x There is a NAT between the internal server IP and the PDA, and there is a nat between internal IP and Internet. Do you have host=dynamic in your * sip.conf entry for 701 ? Actually might help to toss your sip.conf entry out here for 701 without the secret. [701] ; Test phone 701 type=friend username=701 secret=very_secret nat=yes host=dynamic context=test_phone canreinvite=yes disallow=all allow=ulaw allow=alaw dtmfmode=rfc2833 qualify=1000 [EMAIL PROTECTED] pickupgroup=1 qualify=yes Do you see any connection attempts on the console? (ie starting * with -gcvv) No, not at all!! bye Ronald Your not far off.. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Sunday, March 06, 2005 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SJphone on PDA registering with Asterisk??? C. Tomlinson wrote: Ronald, You will need to give *more* information than that I have SJphone on my PDA, and have setup a SIP account on *, and it works fine :-) I take it you have setup sjphone to register to *. I take it your PDA has a network connection? I have setup a sip account at asterisk (701:password) I have an asterisk (voip.elmit.com with an IP address) I have setup a new profile on the PDA sip-elmit: Initialization: as suggested Sip proxy: Proxy domain: my IP address Port 5060 Userdamain: voip.elmit.com Advanced options (nothing set) Sip: Expose software version Enable STUN unsage Redirection: nothing selected STUN: as suggested Use elimit-sip elmit-sip in use (save changes) Display shows: elmit-sip SIP: registering as sip:[EMAIL PROTECTED] ... Host address: 192.168.1.101 NAT/Firewall: Full Cone NAT -- Ronald (office) (Ro) sip:[EMAIL PROTECTED] click on dial Nothing happens, .. not registered in *, ... What have I done wrong? bye Ronald -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SJphone on PDA registering with Asterisk???
All of the SIP and IAX users can register with the server and can send/receive calls. It's just the ISDN side of things that doesn't seem to work. It looks like the ISDN line is functioning because it can reach the server and make a request to create a pipe. So it could be some sort of configuration problem on the routing side or with the softphones themselves... I duno!! Another question... would there be some configuration problem with the X-Lite softphone? For example does it need to be setup in any particular way to receive/send calls to the ISDN line using msn's? Just a thought! Cheers, Philip Lee BTexact *01473 648158 *[EMAIL PROTECTED] * pp 101D Gemini Buildings Adastral Park (SST-MH) Martlesham Heath Ipswich Suffolk IP5 3RE BTexact Technologies is a trademark of British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 180 This electronic message contains information from British Telecommunications plc which may be privileged or confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the numbers or address above) immediately. Activity and use of the British Telecommunications plc E-mail system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Pooton Sent: 07 March 2005 16:23 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SJphone on PDA registering with Asterisk??? Well, given your setup and the fact that you aren't seeing anything on the console with verbose debugging on, I'm going to guess there is a network/routing issue here. I'd try getting the PDA on line and just doing some simple ping tests to the 192.168.250.x network from it. (including to the * server). If you can reach it then is surely should let you register or at least give you info on the console. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Sunday, March 06, 2005 8:59 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SJphone on PDA registering with Asterisk??? James Pooton wrote: I'm all so using SJphone on my x50v, works surprisingly well :). Is voip.elmit.com also in the 192.168.1.X NAT space that your PDA is in? There might be the problem: I have the server at two ethernet cards reachable: Extern with a public IP Intern with 192.168.250.20 on this internal LAN is a wireless accesspoint, which in return changes the IP address to a network 192.168.1.x There is a NAT between the internal server IP and the PDA, and there is a nat between internal IP and Internet. Do you have host=dynamic in your * sip.conf entry for 701 ? Actually might help to toss your sip.conf entry out here for 701 without the secret. [701] ; Test phone 701 type=friend username=701 secret=very_secret nat=yes host=dynamic context=test_phone canreinvite=yes disallow=all allow=ulaw allow=alaw dtmfmode=rfc2833 qualify=1000 [EMAIL PROTECTED] pickupgroup=1 qualify=yes Do you see any connection attempts on the console? (ie starting * with -gcvv) No, not at all!! bye Ronald Your not far off.. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Sunday, March 06, 2005 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SJphone on PDA registering with Asterisk??? C. Tomlinson wrote: Ronald, You will need to give *more* information than that I have SJphone on my PDA, and have setup a SIP account on *, and it works fine :-) I take it you have setup sjphone to register to *. I take it your PDA has a network connection? I have setup a sip account at asterisk (701:password) I have an asterisk (voip.elmit.com with an IP address) I have setup a new profile on the PDA sip-elmit: Initialization: as suggested Sip proxy: Proxy domain: my IP address Port 5060 Userdamain: voip.elmit.com Advanced options (nothing set) Sip: Expose software version Enable STUN unsage Redirection: nothing selected STUN: as suggested Use elimit-sip elmit-sip in use (save changes) Display shows: elmit-sip SIP: registering as sip:[EMAIL PROTECTED] ... Host address:
[Asterisk-Users] anybody tried Fujitsu-Siemens PRIMERGY RX200 S2 server width te4xx?
Hi,
anybody has experience with ${subject} server (intel E7520 based)?
I red some incompatible problems with new intel mb chipsets and digium
cards, but I don't remember which chipsets on black list.
A
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[Asterisk-Users] CAPI questions
Hi all, I have two questions regarding CAPI. Excuse the fact that they are very 'newbie' in nature, but the CAPI documentation is wafer thin! Firstly I have four BRI adapters (all trunks and controlled by CAPI) in my * box and I would like to know whether I can group these together for dialling out in the same way that ZAP channels can be grouped together. Secondly I have a problem where * doesn't seem to recognise incoming calls when one of the B channels is in use. If someone is on the phone to an external number, for example, then incoming calls ring (for the caller, at least) but * doesn't seem to have any idea that the channel is ringing. Lastly, my capi.conf (as below) only defines one controller as this is what we are testing with. My understanding is that the interface block (starting with 'msn=470' and ending with 'devices=2') needs to be repeated for each of the four BRI adapters, but with the correct MSN for each. The documentation I have seen is ambiguous, can anyone confirm this is correct? Thanks in advance, I M Newbie. ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 musiconhold=random [interfaces] msn=470 incomingmsn=* controller=1 softdtmf=1 accountcode= context=incoming ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 -- FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Im a noob
A common issue you will have with FXO/PSTN lines with sip is echo. Test thoroughly before you go live and have 20 people yelling, I can hear myself talk. On Fri, 2005-03-04 at 15:39, Ty Purcell wrote: Yes it does support a basic analog line (or many many lines...). It also supports T1's, ISDN, etc. FXO would provide an analog connection to the phone company (your wall jack) FXS would allow you to plug analog phones into Asterisk. Phone ---(FXS)---Asterisk(FXO)Phone Company You could eliminate the FXS need if you run SIP or IAX IP handsets. Then they would just connect to your network. Ty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, March 04, 2005 3:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Im a noob Im completly new to the whole PBX thing. I have a toshiba unit now and we're moving our office in the next few months. I want to use asterisk but would like to test it out first. Does it support a basic analog phone line like the one in my house? Is that FXS? Are there any FAQs I should read to learn more? Thanks for the reply! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hardphone deployment recommendation
On Mon, March 7, 2005 10:53 am, Robinson Tim-W10277 said: I had the same issue with 'initialising network.' I had a very fast response back from their support desk. Solution is to unplug the network cable reset back to factory defaults from the keypad menu. Tried that (and again just now to be sure) and no luck. I have the RMA so it will go back today. Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 setvars help needed
I'm trying to pass a variable between servers using setvar in iax.conf.
I have a box (ts2) with a t100p in it. It answers the call and dials
another box (ast0) via IAX. I want to pass a variable along with the call
from ts2 to ast0.
I'm running CVS-HEAD-03/07/05 on ts2 and ast0.
ts2's iax.conf:
[general]
disallow= all
allow = ulaw
[ast0]
host= ast0
setvar=foo=bar
type= friend
ts2's extensions.conf:
[ani-block]
exten = _.,1, noop(${CONTEXT}:${EXTEN}:${PRIORITY})
exten = _.,n, answer
exten = _.,n, resetcdr(w)
exten = _.,n(ani-block),agi(ani-block)
exten = _.,n, dial(iax2/ts2:[EMAIL
PROTECTED]/${EXTEN})
exten = _.,n, hangup
exten = _.,ani-block+101, background(vm-sorry)
exten = _.,n, hangup
exten = h,1,hangup
exten = i,1,hangup
exten = t,1,hangup
ast0's iax.conf:
[ts2]
auth= plaintext
context = main
host= ts2
secret = xx
type= friend
username= ts2
ast0's extensions.conf:
[main]
exten = h,1,hangup
exten = i,1,goto(${CONTEXT},${DNIS},1)
exten = t,1,hangup
exten = _.,1, noop(${CONTEXT}:${EXTEN}:${PRIORITY})
exten = _.,n, answer
exten = _.,n, noop(${FOO})
exten = _.,n, noop(${foo})
exten = _.,n, setvar(DNIS=${EXTEN})
exten = _.,n, resetcdr(w)
exten = _.,n, goto(enter-card-number,s,1)
exten = _.,n, hangup
The variable foo is not visible in ast0.
Any clues will be greatly appreciated :)
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline [EMAIL PROTECTED]Fax: +1-760-731-3000
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[Asterisk-Users] 3COM 3101 SIP
I have been (un?)lucky enough to be given a 3COM 3101 phone as a demo to play with and see if I can get it to work with ASTERISK. Supposedly it is SIP, but there is absolutely no documentation with the phone and it doesn't seem to have very many programmable options. 3COM doesn't seem to have any information on their knowledge base about this particular phone. Has anyone had any luck getting one of these to work? It's a nice looking little phone, but so far that's my entire assessment. I am at a loss as to how to get Asterisk to recognize it since it doesn't seem to allow for me to set a username, secret, or for that matter anything more than an IP address. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk for Live-Stream?
At Sat, 05 Mar 2005 20:19:06 +0100, Philipp von Klitzing wrote: How about icecast: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices Another approach: Dial into a MeetMe conference, and connect some client to that conference that takes care of the streaming part. I used Icecast, but without the conferencing part: exten = 9779619,1,Ices(/home/feklee/asterisk/asterisk-ices.xml) It works fine! -- Felix E. Klee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a way to find free zap channels on remote servers ??
how about using chanisavail via manager api On Thu, 2005-03-03 at 16:21, Paco Perez wrote: Hello: I would like to know if there's a way to request free chanels from remote asterisk servers ? My idea is to make an agi returning a dial to inter-asterisk connected servers when there's not enought chanels on local server, maybe like a ping to all of them or maybe requesting to a central server where all the *s send and request information about available chanels each 2 or 3 seconds, it has not about dial plans because I make LCR first and I have a flat rate for national calls, It is about using less analog lines with constant costs every month. Maybe Asterisk has internals for manage this situation (like virtual group of different asterisks chanels) But I would like to be sure 90% that a free chanel is going to be available when I dial to another asterisk and not to have calls rounding over Internet. Thanks for your comments. Paco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF to Email
I need some suggestions (not necessarily using Asterisk?) on how to accomplish the following in the easiest way possible. I would like to have a ~3 prompt VM system, that would ask for some numbers from a caller (case number, id and another id). It would then take their DTMF presses and format an email to a predetermined address (i.e. the email always goes to the same place). I don't really care about the format so much as long as the emails are the same. I.E Lets say I enter VM Prompt 1: 2004123441 VM Prompt 2: 5955 VM Prompt 3: 34 It would format email and send it out something like: To:[EMAIL PROTECTED] Subject: 2004123441 5955 34 Anyone know of something cheap and easy to handle this problem? TIA, --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI questions
Lastly, my capi.conf (as below) only defines one controller as this is what we are testing with. My understanding is that the interface block (starting with 'msn=470' and ending with 'devices=2') needs to be repeated for each of the four BRI adapters, but with the correct MSN for each. If you have different MSN then you have to repeat it for each controller. If they are on the same MSN you can enter devices=8 and controller=1,2,3,4 or repeat which should also work. Elmar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help needed
On 12:41, Mon 07 Mar 05, Alistair Cunningham wrote: snip If neither of the above are possible, consider using IP takeover. This is a tricky thing to make work 100% - you may need expert help. I got this up and running with CARP in half a day. Learned PF, OpenBSD installation and CARP setup. It really isn't that hard anymore. /snip -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF to Email
Asterisk with a simple agi routine could do this easily. Jonathan - Original Message - From: Eric Balsa [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: March 7, 2005 1:02 PM Subject: [Asterisk-Users] DTMF to Email I need some suggestions (not necessarily using Asterisk?) on how to accomplish the following in the easiest way possible. I would like to have a ~3 prompt VM system, that would ask for some numbers from a caller (case number, id and another id). It would then take their DTMF presses and format an email to a predetermined address (i.e. the email always goes to the same place). I don't really care about the format so much as long as the emails are the same. I.E Lets say I enter VM Prompt 1: 2004123441 VM Prompt 2: 5955 VM Prompt 3: 34 It would format email and send it out something like: To:[EMAIL PROTECTED] Subject: 2004123441 5955 34 Anyone know of something cheap and easy to handle this problem? TIA, --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI questions
Thanks Elmar. I assume it is up to the carrier to determine the MSN for each connection? D. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Elmar Haneke wrote: Lastly, my capi.conf (as below) only defines one controller as this is what we are testing with. My understanding is that the interface block (starting with 'msn=470' and ending with 'devices=2') needs to be repeated for each of the four BRI adapters, but with the correct MSN for each. If you have different MSN then you have to repeat it for each controller. If they are on the same MSN you can enter "devices=8" and "controller=1,2,3,4" or repeat which should also work. Elmar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tweaking AGGRESSIVE_SUPPRESSOR
Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while they are talking. I haven't been able to find anywhere in the MEC2 source to limit when it mutes the remote party. It seems to do it with just the slightest bit of sound coming from the room. What are my options besides getting mad and ordering a PRI and a TE100? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tweaking AGGRESSIVE_SUPPRESSOR
On March 7, 2005 01:23 pm, Dennis Webb wrote: Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while they are talking. I haven't been able to find anywhere in the MEC2 source to limit when it mutes the remote party. It seems to do it with just the slightest bit of sound coming from the room. What are my options besides getting mad and ordering a PRI and a TE100? PRI and TE110 won't save you; we've had echo issues with our TE405P and a Bell Canada PRI. All the PRI does is ensure *you* are not causing echo. Now I'm curious -- What physical phones are on either side of this call? Do you have a speakerphone on? I've never heard of an echo canceller acting how you describe, but lots of speakerphones do exactly that. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] working system for months suddenly stopped today with Failed to authenticate on INVITE to
I am getting a log message of Failed to authenticate on INVITE to ... after months of a system working. I have changed nothing... What can cause this. I did some searching and tried setting in sip.conf (canreinvite to both yes and no - made no difference) by default I had no entry at all when this started happening. I am using sip phones, grandstream, cisco combination and all running sip. Calls can come in just fine. just cant make any calls out. Whey trying I get the Failed to authenticate. Any ideas? Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tweaking AGGRESSIVE_SUPPRESSOR
This seems to be how AGGRESSIVE_SUPPRESSOR works. To make sure you don't get echo, it does what a speakerphone does, mute the other party if it hears audio from your end. There is a setting in mec2_const.h for AGGRESSIVE_HCNTR=160 that says in the comments 20ms, I'm assuming this is to tell how long to suppress the other party. There is nothing on this that I have found anywhere and since we are live, I can't change until later to see how it works. We have Polycom SIPS for users, and it doesn't matter what the other party is. It seems from another thread, that the problem midcall is that the electrical properties of the line change midcall causing the echo to return. Without AGGRESSIVE_SUPPRESSOR defined the first minute or so is fine, then a click happens and the echo begins. The phones also seem to go extra sensitive then. You can then hear even keyboard clicks from typing where you don't normally. I've wondered if it's the zaptel cards or poor electricity at my place to the asterisk server. I have put in a SmartUPS 1500 to try to condition electricity there just to make sure. As far as echo and PRI, thanks for making me cry since I just knew that would solve it. On Mon, 2005-03-07 at 12:31, Andrew Kohlsmith wrote: On March 7, 2005 01:23 pm, Dennis Webb wrote: Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while they are talking. I haven't been able to find anywhere in the MEC2 source to limit when it mutes the remote party. It seems to do it with just the slightest bit of sound coming from the room. What are my options besides getting mad and ordering a PRI and a TE100? PRI and TE110 won't save you; we've had echo issues with our TE405P and a Bell Canada PRI. All the PRI does is ensure *you* are not causing echo. Now I'm curious -- What physical phones are on either side of this call? Do you have a speakerphone on? I've never heard of an echo canceller acting how you describe, but lots of speakerphones do exactly that. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting up asterisk with current PBX?
We currently have a Toshiba Perception EX and I would like to start moving toward VOIP. Is there anyway we can run these two systems in parrallel? Better yet, is there anyway we can run fully on asterisk using the current PBX hardware? The current PBX has a mix of analog, digital and electronic cards in it. I tried to google for advice but I didn't find anything that pertained to this. -Thanks Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tweaking AGGRESSIVE_SUPPRESSOR
Andrew Kohlsmith wrote: On March 7, 2005 01:23 pm, Dennis Webb wrote: Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while they are talking. I haven't been able to find anywhere in the MEC2 source to limit when it mutes the remote party. It seems to do it with just the slightest bit of sound coming from the room. What are my options besides getting mad and ordering a PRI and a TE100? PRI and TE110 won't save you; we've had echo issues with our TE405P and a Bell Canada PRI. All the PRI does is ensure *you* are not causing echo. Now I'm curious -- What physical phones are on either side of this call? Do you have a speakerphone on? I've never heard of an echo canceller acting how you describe, but lots of speakerphones do exactly that. What he describes is echo suppression. Because an echo canceller can, generally, only remove some part of an echo, not the entire echo, systems are generally designed to suppress the residual echo in some circumstances. Old speakerphones had poor on no echo cancellation, so the suppression kicked in like that, because it was the only choice. In modern systems, the echo cancellation is much better, so suppression is not needed as much, and when it is used, it's probably done much more imperceptibly (with comfort-noise and stuff like this). The AGGRESSIVE_SUPPRESSOR option enables, as it is named, more aggressive echo suppression. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Setting up asterisk with current PBX?
Welcome to the wiki located here... http://www.voip-info.org/wiki-Asterisk Also, refine your google search to include this at the beginning... Site:lists.digium.com That tells Google, to search only the pages from this email list. Regards, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Hawthorne Sent: Monday, March 07, 2005 11:50 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Setting up asterisk with current PBX? We currently have a Toshiba Perception EX and I would like to start moving toward VOIP. Is there anyway we can run these two systems in parrallel? Better yet, is there anyway we can run fully on asterisk using the current PBX hardware? The current PBX has a mix of analog, digital and electronic cards in it. I tried to google for advice but I didn't find anything that pertained to this. -Thanks Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up asterisk with current PBX?
Thanks for the exceptionaly fast response. I got all the info I need now! On Mon, 7 Mar 2005 11:56:55 -0700, Wiley Siler [EMAIL PROTECTED] wrote: Welcome to the wiki located here... http://www.voip-info.org/wiki-Asterisk Also, refine your google search to include this at the beginning... Site:lists.digium.com That tells Google, to search only the pages from this email list. Regards, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Hawthorne Sent: Monday, March 07, 2005 11:50 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Setting up asterisk with current PBX? We currently have a Toshiba Perception EX and I would like to start moving toward VOIP. Is there anyway we can run these two systems in parrallel? Better yet, is there anyway we can run fully on asterisk using the current PBX hardware? The current PBX has a mix of analog, digital and electronic cards in it. I tried to google for advice but I didn't find anything that pertained to this. -Thanks Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Setting up asterisk with current PBX?
No worries. You are in for a treat as Asterisk is a killer app that can do many things. The thing to remember is that it will take some reading and testing to get things the way you want so don't get discouraged when you have to read a billion pages and search the internet for answers. When you do get it setup right, it works great and is far cheaper than traditional PBX systems. Hope you stick with it, succeed in your Asterisk education and setup, and that someday I see you on the list as a contributor. Best regards, Wiley -Original Message- From: Jason Hawthorne [mailto:[EMAIL PROTECTED] Sent: Monday, March 07, 2005 12:08 PM To: Wiley Siler Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Setting up asterisk with current PBX? Thanks for the exceptionaly fast response. I got all the info I need now! On Mon, 7 Mar 2005 11:56:55 -0700, Wiley Siler [EMAIL PROTECTED] wrote: Welcome to the wiki located here... http://www.voip-info.org/wiki-Asterisk Also, refine your google search to include this at the beginning... Site:lists.digium.com That tells Google, to search only the pages from this email list. Regards, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Hawthorne Sent: Monday, March 07, 2005 11:50 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Setting up asterisk with current PBX? We currently have a Toshiba Perception EX and I would like to start moving toward VOIP. Is there anyway we can run these two systems in parrallel? Better yet, is there anyway we can run fully on asterisk using the current PBX hardware? The current PBX has a mix of analog, digital and electronic cards in it. I tried to google for advice but I didn't find anything that pertained to this. -Thanks Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple outside phones
Is there anyway to have multiple VOIP phones (from inside NAT firewalls and not) connect to my single * server? What do I need? I could put my * server on the outside of the my firewall but I'd rather not. Does the SIP Express server help at all? I can get phones to connect but I dont get any voice. I'm assuming it's NATn issues. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: What my IAXy could have been...
http://www.farfon.com/ http://ipphone.eezeephone.com/ Looks like all URLs on IAX-capable phones, http://www.iaxtalk.com/ included, point to China. Interesting... Farfon is in Pakistan, not China ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] multiple outside phones
Check the can reinvite setting for NAT issues. Check the wiki for how to configure as you have described. http://www.voip-info.org/tiki-index.php?page=Asterisk Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, March 07, 2005 1:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] multiple outside phones Is there anyway to have multiple VOIP phones (from inside NAT firewalls and not) connect to my single * server? What do I need? I could put my * server on the outside of the my firewall but I'd rather not. Does the SIP Express server help at all? I can get phones to connect but I dont get any voice. I'm assuming it's NATn issues. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to get (cheap) VoIP
I would like to deploy a (very) small PBX at my place, so that I can stop answering phones for my kids or my wife, using distinctive ringings. Why not just buy a phone capable of distinctive ringing? I think Siemens makes a few for example? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk MySQL Blobs
Hello Folks, Has anyone had production experience using * w/ MySQL Blobs to store sound files? The application I am working on requires all user data resides in a database. I am currently reading/writing the files to disk via a phpagi scripts but I would love to read the blob into a variable in the dial plan, etc. It seems like a waste of resources to write and delete the file. Thanks, Vinko Grskovic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup
Thanks for the suggestion. I have busydetect=yes in zapata.conf. You refer to Martin's algorithm. Can you provide more details please? - Original Message - From: Soner Tari [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup detection failing To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 If your problem is the same as mine then you need to use busydetect. In Turkey, we don't have polarity reversal, and signalling tones are quite different. But just enabling busydetect in zaptel.conf did not help me, it may work for you though. I had to change relevant compile options and some settings in dsp.c file and recompile. (The goal is to detect congestion tone here.) I haven't noticed any problems so far, but I am still working on it to use Martin's algorithm instead. Hope this helps, Soner - Original Message - From: Cameron Beattie To: asterisk-users@lists.digium.com Sent: Monday, March 07, 2005 1:31 AM Subject: [Asterisk-Users] FXO module in TDM400P (UK,BT) - Hangup detection failing I am based in New Zealand and am experiencing the same problem as referred to in the post FXO module in TDM400P (UK, BT) - Hangup detection failing from 2 November 2004 i.e. Zap/4 (being the FXO module) not detecting hangup on the PSTN line if the call is not answered on a PABX extension. Has anyone managed to find a resolution to the problem? For information: Digium TDM400P with FXS on Zap 1 2 and FXO on Zap 3 4. CVS-v1-0-01/24/05 Using fxs_ks signalling Regards Cameron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] multiple outside phones
canreinvite is no for both phones (internal and the one external) Wiley Siler [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 03/07/2005 03:03 PM Please respond to Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com cc Subject RE: [Asterisk-Users] multiple outside phones Check the can reinvite setting for NAT issues. Check the wiki for how to configure as you have described. http://www.voip-info.org/tiki-index.php?page=Asterisk Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, March 07, 2005 1:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] multiple outside phones Is there anyway to have multiple VOIP phones (from inside NAT firewalls and not) connect to my single * server? What do I need? I could put my * server on the outside of the my firewall but I'd rather not. Does the SIP Express server help at all? I can get phones to connect but I dont get any voice. I'm assuming it's NATn issues. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BroadVoice configuration changes for Outbound
Doing this with no notification whatsoever, let alone notification sufficiently in advance of these changes, was stupid and careless. This move probably broke a significant number of your customers' telephones service. One can only guess at the impact that this careless move had on your customer service department. In the future, give some thought to planning such changes more carefully, announcing them well in advance of implemenatation. I am satisfied enough with my BroadVoice service that I will overlook this incident, but there are lots of other vendors out there. Surely, at least one of them has more concern for their customers than BroadVoice has demonstrated with this fiasco. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Weber Sent: Saturday, March 05, 2005 9:13 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] BroadVoice configuration changes for Outbound Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=phonenumber authuser=phonenumber secret=registration password Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.0 - Release Date: 03/02/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.0 - Release Date: 03/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Im a noob
That is true - I've run into it on some of my polycoms. After tweaking the phone's built-in echo cancellation I was able to eliminate it though. Ty -Original Message- From: Dennis Webb [mailto:[EMAIL PROTECTED] Sent: Monday, March 07, 2005 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Im a noob A common issue you will have with FXO/PSTN lines with sip is echo. Test thoroughly before you go live and have 20 people yelling, I can hear myself talk. On Fri, 2005-03-04 at 15:39, Ty Purcell wrote: Yes it does support a basic analog line (or many many lines...). It also supports T1's, ISDN, etc. FXO would provide an analog connection to the phone company (your wall jack) FXS would allow you to plug analog phones into Asterisk. Phone ---(FXS)---Asterisk(FXO)Phone Company You could eliminate the FXS need if you run SIP or IAX IP handsets. Then they would just connect to your network. Ty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, March 04, 2005 3:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Im a noob Im completly new to the whole PBX thing. I have a toshiba unit now and we're moving our office in the next few months. I want to use asterisk but would like to test it out first. Does it support a basic analog phone line like the one in my house? Is that FXS? Are there any FAQs I should read to learn more? Thanks for the reply! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk MySQL Blobs
Has anyone had production experience using * w/ MySQL Blobs to store sound files? The application I am working on requires all user data resides in a database. I am currently reading/writing the files to disk via a phpagi scripts but I would love to read the blob into a variable in the dial plan, etc. It seems like a waste of resources to write and delete the file. Too bad your requirement is to have everything in the DB, 'cause you will be asking for trouble in the long run. BLOBs are probably the fastest way to kill your DB once you scale. I did an experiment a few years ago to stream faxes as BLOB's into a SQL server and performance beyond a few thousand records was to put it mildly crap. IMO, use filesystem for files. Use DB for DB. Put a pointer in a field to the file. Your DB will love you for it. WinFS is the Microsoft solution to this problem (assuming it ever ships and gets backported), but I think the Linux guys are doing something like it with Reiser4, there's a plug in for this?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MySQL Blobs
Checkout CVS. There is now support for storing voicemail sound files in DB with ODBC. -Matthew - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 07, 2005 2:05 PM Subject: [Asterisk-Users] Asterisk MySQL Blobs Hello Folks, Has anyone had production experience using * w/ MySQL Blobs to store sound files? The application I am working on requires all user data resides in a database. I am currently reading/writing the files to disk via a phpagi scripts but I would love to read the blob into a variable in the dial plan, etc. It seems like a waste of resources to write and delete the file. Thanks, Vinko Grskovic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 600 XML
Hey guys, Im interested in the XML Support that the Polycom phones have. I want my techs to be able to view queue information via the XML screen. Is this possible? When I say queue information, I mean how many people are waiting in the queues (3 queues combined), how long the wait is for new call-ins etc. Things like that. Is this possible? Please let me know guys! Thanks a lot for all of your help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: What my IAXy could have been...
Well let's try to figure this out. 1. The biggest telecommunications market in the world (at least mobile according to the latest reports, 125,000,000 mobile users). 2. One of the countries in this world where calling long distance is still a luxury, so VOIP is almost a household item. Correct me if I'm wrong. On Mon, 7 Mar 2005 21:00:40 +0100, Wilson Pickett [EMAIL PROTECTED] wrote: http://www.farfon.com/ http://ipphone.eezeephone.com/ Looks like all URLs on IAX-capable phones, http://www.iaxtalk.com/ included, point to China. Interesting... Farfon is in Pakistan, not China ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial, record, save to voicemail
I want Asterisk to do the following:
- call a voicemail system by dialing a number and
playing a DTMF tone
- record what is said by the called party and save
the recording to a file
- end the recording when a particular phrase is
said by the called party
- put that recording into an Asterisk voicemail box
and notify the user
I've made a start below (on the easy bit). Any
further pointers on how to proceed would be greatly appreciated.
[macro-callminder_retrieve]
exten =
s,1,Dial(${LOCALTRUNK},083210,6,D(www1))
;this dials the callminder number, waits and then
plays DTMF tone 1 to retrieve new messages
exten =
s,2,Record(/tmp/msg000%d.gsm|0|20)
;this doesn't seem to work
For those who wonder why: We have a small business
with two incoming lines. We have "Call Minder" where the incoming calls are
recorded to voicemail if both lines are busy. By implementing Asterisk we will
have two voicemail boxes, the Asterisk one (where calls aren't answered) and the
Call Minder one (when lines are busy). The idea is to retrieve the messages from
Call Minder on a regular basis and put them into the Asterisk voicemail
box.
Regards
Cameron
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Re: [Asterisk-Users] 3COM 3101 SIP
On Mon, 7 Mar 2005, PA wrote: I have been (un?)lucky enough to be given a 3COM 3101 phone as a demo to play with and see if I can get it to work with ASTERISK. Supposedly it is SIP, but there is absolutely no documentation with the phone and it doesn't seem to have very many programmable options. 3COM doesn't seem to have any information on their knowledge base about this particular phone. Has anyone had any luck getting one of these to work? It's a nice looking little phone, but so far that's my entire assessment. I am at a loss as to how to get Asterisk to recognize it since it doesn't seem to allow for me to set a username, secret, or for that matter anything more than an IP address. A while back (quite a while actually) someone on the list mentioned that the 3com phones download their program from the 3com pbx every time they power up. Without the program they are unable to operate. All this is second- or third hand information. Once bootstrapped they may be regular sip phones. Try searching the list for 3com 3102. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 600 XML
Ken Sandell wrote: Hey guys, Im interested in the XML Support that the Polycom phones have. I want my techs to be able to view queue information via the XML screen. Is this possible? When I say queue information, I mean how many people are waiting in the queues (3 queues combined), how long the wait is for new call-ins etc. Things like that. Is this possible? Please let me know guys! Thanks a lot for all of your help. Ken, First things first: this is not a -dev question, and you really shouldn't cross post... Anyways, it should not be that hard to write a CGI app that can connect to the Asterisk Manager and output the information that you are looking for in XHTML. I would take a look at the Wiki for any Manager stuff (including PHP Perl code samples) and then check out some sites that provide XHTML. Between the two you should get something that works. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 600 XML
Hi, Yes, is possible. I use my XML browser for that. Best regards, Chris HARIGA From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Sandell Sent: Monday, March 07, 2005 3:57 PM To: asterisk-dev@lists.digium.com; asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP 600 XML Hey guys, Im interested in the XML Support that the Polycom phones have. I want my techs to be able to view queue information via the XML screen. Is this possible? When I say queue information, I mean how many people are waiting in the queues (3 queues combined), how long the wait is for new call-ins etc. Things like that. Is this possible? Please let me know guys! Thanks a lot for all of your help. smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MySQL Blobs
Hi Vinko, MySQL blobs will store binary data, so you should be OK there. I'd focus on whether or not storing the data in a variable is a good idea. Typically, with any programming language, it's good practice to keep variable lengths short so you aren't passing the variable itself between functions. I can't say if that could cause performance issues under higher load. I'd love to hear how you make out, as well as anyone else's input. - Eric On Mon, 07 Mar 2005 15:05:32 -0500 [EMAIL PROTECTED] wrote: Hello Folks, Has anyone had production experience using * w/ MySQL Blobs to store sound files? The application I am working on requires all user data resides in a database. I am currently reading/writing the files to disk via a phpagi scripts but I would love to read the blob into a variable in the dial plan, etc. It seems like a waste of resources to write and delete the file. Thanks, Vinko Grskovic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
