Re: [asterisk-users] 1.6b9 Audio Issue
To close the loop on this I have found that this appears to no longer be an issue since I moved to 1.6rc6. Mark Michelson wrote: MFH wrote: I'm noticing in 1.6 Beta 9 that on outgoing calls I get a brief audio drop when the audio starts on the other end of the call. So basically I hear the first word, miss the second word and then hear the rest fine. I've noticed this after calling multiple locations and getting some recording on the other end. The origin of the outbound channel is always SIP but the asterisk to PSTN could be SIP or IAX. Anyone else? MARK. One difference between Asterisk 1.6.0 and previous versions is that when a channel answers, there is a built-in 500 ms delay so that media has time to be set up. This may be what you are experiencing. There was a bug reported recently that was traced back to this delay. In the next 1.6.0 tarball, the delay will behave slightly differently, although I doubt it will be noticeable for the situation you have described. The bug I refer to is: http://bugs.digium.com/view.php?id=12924 Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Fedora 9
The best way I can think of is: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz tar -zxvf asterisk-1.4.21.2.tar.gz cd asterisk-1.4.21.2 ./configure make menuselect (You don't have to select anything) make make install make samples Pascal Bruno wrote: I am about to install Asterisk on a Fedora 9 box, but i see with yum, they only have Asterisk 1.6 beta in the package repos which I didn't really want to install until they have a stable release. Does anybody know or have a good and easy way to install Asterisk 1.4 on fedora 9? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Fedora 9
It's like the same except you wget a different package and I don't think you have a menuselect option and you do it before you compile asterisk. For addons I think there might be some configuration if you are planning to use the database stuff which I don't use. The sounds come with the asterisk install and the menuselect allows you to decide which sounds you want. But, if compiling is foreign to you as someone points out maybe you should not take that route. I don't agree that upgrading is difficult with this process though. Pascal Bruno wrote: Ok very good, how about for the asterisk addonds and sounds? Can you provide me the commands to get, build and install for the 1.4.21 version? Thanks a lot guys. On Fri, Sep 12, 2008 at 6:07 AM, MFH [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The best way I can think of is: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz tar -zxvf asterisk-1.4.21.2.tar.gz cd asterisk-1.4.21.2 ./configure make menuselect (You don't have to select anything) make make install make samples Pascal Bruno wrote: I am about to install Asterisk on a Fedora 9 box, but i see with yum, they only have Asterisk 1.6 beta in the package repos which I didn't really want to install until they have a stable release. Does anybody know or have a good and easy way to install Asterisk 1.4 on fedora 9? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6rc4 chan_iax2 messages
As a result of: http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html I am seeing these messages when upgrading for 1.6b9 to 1.6rc4. Is there something I should be doing to address this warning? [Sep 4 13:31:34] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 14670, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:34:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 13447, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:39:14] WARNING[2956]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 14442, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:40:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 11096, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:42:14] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 14517, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:42:54] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 2761, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:43:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 10117, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:44:54] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 4215, but no such call exists (and I cannot remove lagid, either). MARK. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6rc4 chan_iax2 messages
I was on the call at the time and was not experiencing any apparent problems. As I was responding I did some further investigation and saw the messages even when there wasn't an active call (so I thought). I looked at the active IAX channels: [Sep 4 14:27:14] WARNING[2956]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 6467, but no such call exists (and I cannot remove lagid, either). [Sep 4 14:35:14] WARNING[2958]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 5345, but no such call exists (and I cannot remove lagid, either). asterisk*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None)67.202.54.166(None) 02713/0 1/0 0ms -0001ms ms unknow 1 active IAX channel and determined that this is a connection to an Amazon EC2 image that I was using for testing but recently deactivated: asterisk*CLI iax2 show peers Name/UsernameHost Mask Port Status asterisk2/aster 67.202.54.166 (S) 255.255.255.255 4569 UNREACHABLE [asterisk2] type=friend username=asterisk2 secret=pass auth=md5 host=asterisk2.myhost.com context=frompeer peercontext=frompeer qualify=yes trunk=yes I'm going to remove this definition and see if the messages go away. MARK. Tilghman Lesher wrote: On Thursday 04 September 2008 12:59:33 MFH wrote: As a result of: http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html I am seeing these messages when upgrading for 1.6b9 to 1.6rc4. Is there something I should be doing to address this warning? [Sep 4 13:31:34] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 14670, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:34:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 13447, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:39:14] WARNING[2956]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 14442, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:40:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 11096, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:42:14] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 14517, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:42:54] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 2761, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:43:34] WARNING[2949]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 10117, but no such call exists (and I cannot remove lagid, either). [Sep 4 13:44:54] WARNING[2951]: chan_iax2.c:1200 __send_lagrq: I was supposed to send a LAGRQ with callno 4215, but no such call exists (and I cannot remove lagid, either). They're actually pretty harmless messages. I may wind up moving them to DEBUG only. You'll only get them when a host is unreachable during a call, though, so you may want to figure out why that host became unreachable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX to work on two ports: 4569 and 4570
Depending on how smart your router configuration is, you can leave both boxes A and B on port 4569 and then just set up two rules on your router: Port 4569 - Box A port 4569 Port 4570 - Box B port 4569 If your router is not complex enough to allow a port mapping such as the second line above then you'll need to get box B IAX on port 4570 and I think people have described how to do that. bilal ghayyad wrote: Again, I did not understand. How the router will know if the call that came from Asterisk Box C need to be routed for Asterisk Box A or Asterisk Box B? Note: Asterisk A and B both behind DSL router, while Asterisk C in remote side, both sides are connected via Internet. Regards Bilal -- On Friday 25 July 2008 10:58:28 bilal ghayyad wrote: The reason that I need to do this is: I will have two Asterisk PBX's, and I need both of them to use same Internet (so both of them will be behind NAT under same DSL router), in that case, how I will distinguish on the router the calls that need to be send for box A and the calls that need to send for box B? Asterisk C will be in remote side, and will communicate with Asterisk A and B, so I was look to have two ports on for IAX to be running on Asterisk C, for example: 4569 and 4570, so when box C need to talk with box A, then it sends via 4569 and when it needs to talk with box B then it sends via 4570 port, and DSL router does the routing based on these ports. Is there any other solution? Oh, for outbound traffic, you don't need to define other ports. Simply forward port 4570 on your router to port 4569 on Asterisk B. Remember that unlike SIP, IAX2 does not encode the port number into the protocol, so it'll work perfectly fine this way. If you're bound and determined to listen to port 4570 on Asterisk B, you still don't need to listen on two ports on Asterisk C. Just leave off the port number in iax.conf the specification for Asterisk C on box B. Likewise on Asterisk C, define the specification for Asterisk B to use port=4570. There is no need for port numbers to be the same on both machines when talking IAX. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3-way calling for IAX channels
Asterisk supports conferencing without using meetme. In this case you don't have a central dial in number but a single extension can initiate the conference call. Generally this is done the same way as with traditional PSTN service which is that while on a call between two parties, flash the line, dial out to the third party then flash again and all the parties should be connected. Noah Miller wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommend quality wholesale termination - Singapore and Sydney, Aus
Can anyone recommend decent quality as close to pay-as-you-go SIP wholesale termination providers in both Singapore and Sydney, Australia? I will be in both places and want a local carrier while I'm there. It needs to be easy in and easy out and if it's not $0 base or close I'll need to be able to drop it in a month. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID - Panama
Dean asked for it so he can decide if it's worth it to him but that sounds like the price someone would pay for flatrate and probably not what one would want to pay for 5 calls per day. MARK. Sam Tam wrote: We have got that for $10 USD setup and $25 USD per month If you are interested please email me back Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Friday, July 18, 2008 9:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DID - Panama I need a low monthly rate DID in Panama. Maybe 2-3 inbound calls a day max. 1-2 outbound calls a day only. Will be rarely used but needs to be very good quality and needs longevity of business (eg this number is going into print so company needs to be around for a while). Please email with rates and details. Cheers, Dean /mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6b9 Audio Issue
I'm noticing in 1.6 Beta 9 that on outgoing calls I get a brief audio drop when the audio starts on the other end of the call. So basically I hear the first word, miss the second word and then hear the rest fine. I've noticed this after calling multiple locations and getting some recording on the other end. The origin of the outbound channel is always SIP but the asterisk to PSTN could be SIP or IAX. Anyone else? MARK. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] distinctive ring
It depends on which type of SIP device you have that determines on how you signal a distinctive ring. You need to change the SIP Header like: exten = s,n,SIPAddHeader(Alert-Info:Bellcore-r8) where the number after the 'r' signifies a different ring tone but some devices uses different names other than Bellcore... If you are on an internal path you would set one ring and if you are on an external path set another. Fidel Garcia wrote: This one! The sound of a phone that signals a call coming from internal/external My phones are SIP, I do not know what ZAP means or what it does. Thanks for your reply! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, July 15, 2008 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] distintive ring Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? Fidel, I do not know what kind of tone you mean: The sound of a phone that signals a call coming from internal/external? The sound in the earpiece after you dialled while you wait for the other end to pick up? In the first case distinctive ring is probably the right term to search for. You will have to decide wether your phones are SIP or ZAP (or both, or different), because methods seem to differ. As a start reading point have a look at http://www.malico.com.tw/voip-info/wiki/view/Asterisk+SIP+channels.html The mailing list archives contain a lot of information *hint* Best regards Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.4.11/1553 - Release Date: 7/15/2008 5:48 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] distintive ring
My internal calls start in an entirely different context than calls coming in externally. There's never any confusion about where the call is coming from and I don't use prefixes. Allann Jones wrote: Internal and external calls can be distinguished generally by the phone number. A prefix or the number of digits of the phone number. For example, you could use a digit prefix followed by a interval of time to call a internal number. Examples: Internal number: 0,1234 External number: 87654321 On Tue, Jul 15, 2008 at 2:02 PM, Fidel Garcia [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net -- ___ Allann J. O. Silva I received the fundamentals of my education in school, but that was not enough. My real education, the superstructure, the details, the true architecture, I got out of the public library. For an impoverished child whose family could not afford to buy books, the library was the open door to wonder and achievement, and I can never be sufficiently grateful that I had the wit to charge through that door and make the most of it. (from I. Asimov, 1994) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing inbuilt sound messages
I was curious so I took a look at my sounds directory. Most of the
files are 644 except the g729 which are 444. I also noticed that the
ownerid/groupid are a non-existent 1000/1000. I take it that the sound
installer uses something like tar with the option to keep the original
owner and groupid which it shouldn't be doing. If it's tar it should
use at least the option -o when doing the extraction to
/var/lib/asterisk/sounds.
-rw-r--r-- 1 1000 10006985 Dec 5 2007 zip-code.alaw
-rw-r--r-- 1 1000 10006985 Dec 5 2007 zip-code.g722
-r--r--r-- 1 1000 1000 870 Dec 5 2007 zip-code.g729
-rw-r--r-- 1 1000 10001452 Dec 5 2007 zip-code.gsm
-rw-r--r-- 1 1000 10006985 Dec 5 2007 zip-code.ulaw
-rw-r--r-- 1 1000 1000 14014 Dec 5 2007 zip-code.wav
from asterisk/sounds/Makefile:
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR); cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR)/es; cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR)/fr; cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR); cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR)/es; cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR)/fr; cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst $(MOH_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(MOH_DIR); cat $(PWD)/$${PACKAGE} | gzip -d | tar
xf -) \
Tzafrir Cohen wrote:
On Fri, Jul 11, 2008 at 09:56:29AM +1200, Lists wrote:
I only did the 420 because thats what the original files looked like?
r-- -w- ---
Should I change this to 644?
Yes!
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Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
Very cool, you've piqued my interest. Since I haven't launched an instance before, where's the best place to learn to do that? What's the approximate monthly cost of hosting an Asterisk PBX on EC2? Ronald Lewis wrote: I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. Thanks! Ronald Lewis Denver, Colorado http://ronaldlewis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Call Screener
This is what I use. The Read does have a default timeout but you should
be able to put your own.
extensions.conf:
exten =
s,n(dial),Dial(SIP/sipura2_1SIP/sipura1_1SIP/sipura2_2SIP/spa942_3SIP/aastra480_3,20,mtTM(screen))
exten = s,n(vmail),Voicemail([EMAIL PROTECTED])
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,n,Read(ACCEPT,screen-callee-options,1)
exten = s,n,GotoIf($[${ACCEPT} = 1]?ok:cont)
exten = s,n(ok),Noop
exten = s,s+2(cont),Set(MACRO_RESULT=CONTINUE)
I'm trying to build a simple accept/reject screening app for inbound calls
that * forwards to my cell phone. Basically I want * to announce the caller
ID and then let me press 1 to accept the call or 2 to reject the call and
send the outside party to voicemail.
I've been messing around with variation of the script below... can anyone
tell me what I'm doing wrong? It's got to be something obvious that I've
overlooked.
Thanks!!!
[main]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Wait(1)
exten = s,n,Dial(SIP/[EMAIL PROTECTED],120,gM(screen))
exten = s,n,PlayBack(vm-goodbye)
exten = s,n,Hangup
[macro-screen]
exten = s,1,Wait(1)
;exten = s,n,SayDigits(${CALLERID(num)})
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=30)
exten = s,n,Background(accept-reject)
exten = 1,1,Set(MACRO_RESULT=CONTINUE)
exten = 2,1,PlayBack(vm-goodbye)
exten = 2,2,Hangup
exten = s,6,Wait(10)
exten = i,1,Goto(TT_VO,s,1)
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Re: [asterisk-users] Asterisk as an IVR solution
From what I can tell Read allows for a floating point input which uses ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: Admittedly I have not used the ExternalIVR app. Is it any good? I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, it can do it, but boy it is UGLY. There's also the fact that you can't call Backgound() in a macro, which forces you to use Read() which won't accept a timeout of 1s. There's no DTMF background detection while playing SayDigits so you have to roll your own by calling an external AGI and concatenating sound files. Yuck. By the time you code in logic for handling timeouts and incorrect responses to menu's with all the gotos and what-not, it turns into a god aweful mess. Sure, you can do it. Doug. - Original Message From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 10:37:55 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi. We are building an application that will provide users with the ability to call in and report an absence. The caller will have to validate themselves and the call tree will be dynamic, based on data in a MySQL database. We will have many customers, each calling a separate phone number, each having a different call tree. New customers will be added regularly and we do not want a solution that requires extensive programming each time (the call trees are different in subtle ways from each other). Is Asterisk a great solution for this? If not do you know what would? If so, we need someone to help us set it up, can you suggest someone? Thanks in advance. Best. Mark Asterisk certainly is a great solution for this. If you find you need or want extra flexibility, the external IVR app. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk sip problem
Are asterisk and the phone on the same lan? I see you have nat=no. Do you see the phone adapter registered? Emmanuel Favre-Nicolin wrote: Hi, I'm having a problem to receive inbound call from my sip provider. I used to be OK, I may I have change something (for example I switched from asterisk 1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a configuration problem on my side!) I have basically a sip account and a linksys voip adapter with a phone on it (sip name 1000), configured in asterisk. Outbound call from the phone just work fine. Inbound call fail to ring my phone. When the inbound call occur I see on the asterisk command line : -- Executing [EMAIL PROTECTED]:1] Dial(SIP/callcentric.com-081f1ac8, SIP/1000) in new stack -- Called 1000 -- SIP/1000-081ed5e0 is ringing but my phone is not ringing in sip.conf: [1000] type=friend secret=blablabla qualify=yes; Qualify peer is not more than 2000 mS away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=fromsoftphone port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same host in extensions.conf: [from-callcentric] exten = 17772962667,1,Dial(SIP/1000) exten = 17772962667,n,Hangup() The default extension I got for inbound call is 17772962667 that's why I used that extension. I tu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk sip problem
I don't see anything obvious right away other than have you confirmed that the phone is actually working? Can you get it to ring? With my Sipura adapters that use Linksys software I can view the call status in the Info section which if you have that panel might tell you if the adapter thinks a call is coming in. I just looked at my Info page with a call coming in and I can see the call state as Ringing and a bunch of other details. Call Status Call 1 State: Ringing Call 1 Tone:Ring - Hold Call 1 Encoder: G711u Call 1 Decoder: G711u Call 1 FAX: No Call 1 Type:[L1]Inbound Call 1 Remote Hold: No Call 1 Callback:No Call 1 Peer Name: UNAVAILABLE Call 1 Peer Phone: 1X Call 1 Duration: Call 1 Packets Sent:0 Call 1 Packets Recv:0 Call 1 Bytes Sent: 0 Call 1 Bytes Recv: 0 Call 1 Decode Latency: 0 ms Call 1 Jitter: 0 ms Call 1 Round Trip Delay:0 ms Call 1 Packets Lost:0 Call 1 Packet Error:0 Call 1 Mapped RTP Port: 16420 0 [EMAIL PROTECTED] wrote: They are on the same lan the adapter is registered sip show peers Name/username HostDyn Nat ACL Port Status sippyskypeuser/sippyskype 192.168.2.765070 OK (1 ms) 1000/1000 192.168.2.76 D 5061 OK (1 ms) freephonie-out/0950607456 212.27.52.5 N 5060 OK (766 ms) callcentric/17772962667204.11.192.34N 5080 OK (206 ms) the pap2t's IP is 192.168.2.205 and the IP of the asterisk box is 192.168.2.76 sip show registry HostUsername Refresh State Reg.Time freephonie.net:5060 0950601785 Registered Wed, 09 Jul 2008 10:12:44 callcentric.com:5080177729x 46 Registered Wed, 09 Jul 2008 10:13:29 I use line2 of my pap2t (line 1 is not enabled). Here is the conf : http://emmanuelfavrenicolin.free.fr/Public/Divers/Snapshots1/20080709_pap2t.jpg On 7/9/08, MFH [EMAIL PROTECTED] wrote: Are asterisk and the phone on the same lan? I see you have nat=no. Do you see the phone adapter registered? Emmanuel Favre-Nicolin wrote: Hi, I'm having a problem to receive inbound call from my sip provider. I used to be OK, I may I have change something (for example I switched from asterisk 1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a configuration problem on my side!) I have basically a sip account and a linksys voip adapter with a phone on it (sip name 1000), configured in asterisk. Outbound call from the phone just work fine. Inbound call fail to ring my phone. When the inbound call occur I see on the asterisk command line : -- Executing [EMAIL PROTECTED]:1] Dial(SIP/callcentric.com-081f1ac8, SIP/1000) in new stack -- Called 1000 -- SIP/1000-081ed5e0 is ringing but my phone is not ringing in sip.conf: [1000] type=friend secret=blablabla qualify=yes; Qualify peer is not more than 2000 mS away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=fromsoftphone port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same host in extensions.conf: [from-callcentric] exten = 17772962667,1,Dial(SIP/1000) exten = 17772962667,n,Hangup() The default extension I got for inbound call is 17772962667 that's why I used that extension. I tu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proper Hangup message
It looks like it's 19:
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
Nhadie wrote:
Hi,
How do i send proper message when hanging up?
[from-trunk]
exten = _1234,1,Dial(SIP/${EXTEN}|30|t)
exten = _1234,n,Hangup
With that, the other end receives a call reject if i don't answer the
phone, but the telco said they need something like No Answer instead
of Call Reject.
Is it possible to set that? Thanks
Regards,
Nhadie
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Re: [asterisk-users] Sipura SPA-3102 and Asterisk
You didn't give details of your networking setup but do you have the 3102 and then X-Lite client connected to the same switch or router? It not, one switch could be dropping packets or slow. Do you qualify both devices in Asterisk? Do they have the same ping times? I haven't done any audio streaming so I can't comment on the behavior of that application in various circumstances. David Siegel wrote: I have a Sipura SPA-3102 that I use to connect an analog phone to Asterisk. The analog phone (actually, an analog extension from a Panasonic PBX – but this should not matter) is connected to the LINE1 port of the Sipura. I’ve got the setup working fine, and when I place a call from the phone on LINE1 to an X-Lite soft phone, via Asterisk 1.6, all works fine. Voice quality on the call is perfect. Now, I’ve been playing with MP3Player, using it to stream audio to an extension. If I call the streaming audio extension from my X-Lite soft phone, I hear the mp3 file perfectly. If I call the same extension from the phone on my Sipura LINE1, the sound is very choppy. All these devices are connected on my local area network, so I don’t think there is a networking problem. To me, this is odd, because: - Voice calls placed from the Sipura LINE1 phone work perfectly - MP3 playing from a call placed on an X-Lite client work perfectly - MP3 playing from a call placed on the Sipura LINE1 phone are choppy I’ve played with various settings on the Sipura, with no luck in fixing this issue. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP MWI Problem in 1.4 and 1.6
I've been having a problem with Asterisk MWI notification on my SIP
phones since going to version 1.4 a long time ago. Since going to this
version, I have needed to go into chan_sip.c and do the following:
/*! \brief Check whether peer needs a new MWI notification check */
static int does_peer_need_mwi(struct sip_peer *peer)
{
time_t t = time(NULL);
/* COMMENT OUT THIS CODE
if (ast_test_flag(peer-flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY)
!peer-mwipvt) {
peer-lastmsgcheck = t;
return FALSE;
}
*/
if (!ast_strlen_zero(peer-mailbox) (t - peer-lastmsgcheck)
global_mwitime)
return TRUE;
return FALSE;
}
After commenting out the section noted above then MWI works fine on all
phones. Now, I have moved to 1.6 and am having the same problem except
this function no longer exists. Has anyone had a similar problem and if
so, how did you fix it in your config files assuming it's some setting
in one of them? Does anyone know why this flag test is not working?
Also, if someone knows how I can patch 1.6 to fix this that would be
helpful also. I have the corresponding mailboxes set in my sip.conf
contexts as so:
[sipiura1]
...
[EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED]
subscribemwi=yes
...
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