[asterisk-users] Trouble outgoing VOIP Provider Calls
= _96XX,2,System(mkdir /mnt/data/Recording/${EXTEN:1})
exten =
_96XX,3,Set(CALLFILENAME=/mnt/data/Recording/${EXTEN:1}/${EXTEN:1}-Recei
ved-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${CALLERID(num)}.wav49)
exten = _96XX,4,MixMonitor(${CALLFILENAME}|v(0)V(0))
exten = _96XX,5,Dial(SIP/${EXTEN:1},45,Ttr)
exten = _96XX,6,Voicemail(u${EXTEN:1})
exten = _96XX,7,Hangup
exten = _96XX,106,Voicemail(b${EXTEN:1})
exten = _96XX,107,Hangup
--- Sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
videosupport=yes
context = from-sip
disallow = all
allow = ilbc
allow = ulaw
allow = alaw
nat=yes
srvlookup=no
externip=YYY.YY.YY.YY
localnet=192.168.1.0/255.255.0.0
subscribecontext = sip
maxexpirey=3600
defaultexpirey=600
; Main VOIP Account Register and Secondary 100 Number block Registration
register = 073...:password@byo.engin.com.au/073...
register = 073...:password@byo.engin.com.au/073...
[acevoip]
context=from-acevoip
type=friend
auth=md5
canreinvite=no
dtmfmode=rfc2833
fromdomain=voice.mibroadband.com.au
fromuser=073...
host=byo.engin.com.au
insecure=invite
musiconhold=framed
nat=yes
port=5060
qualify=no
realm=mobileinnovations.com.au
canreinvite=yes
secret=password
username=073...
annexb=no
disallow=all
allow=g729
[610]
type=friend
secret=password
host=dynamic
callerid=James - Office 610
defaultip=192.168.1.230
disallow=all
allow=g729
[EMAIL PROTECTED]
port=5060
dtmfmode=auto
canreinvite=no
call-limit=1
___
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Re: [asterisk-users] TDM2400 Hardware Echo Cancel
Had the exact same issue with the hardware canceller. If I set echocancel=no then the problem goes away. Very weird that the static only happens on our side and we only hear it. - Original Message - From: Webster, Andrew [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 23, 2007 2:42 PM Subject: RE: [asterisk-users] TDM2400 Hardware Echo Cancel I have been having the same problems since installing a TDM2400 with hardware echo canceller. The best way to describe the sound is a background crackle or hiss that just can't be filtered out. Increasing the RX gain just makes the problem worse. SIP to SIP calls are flawless. An acquaintance told me the analog line level is too low, but when plugging a regular phone into the line, the signal is plenty loud enough. I am curious if anyone else had similar issues with the TDM2400 card and if they have resolved it. -- Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Sharples Sent: Tuesday, January 16, 2007 09:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM2400 Hardware Echo Cancel Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + door opener
Seems these two are closer to what's asked for. http://www.vikingelectronics.com/products/view_product.php?pid=343 http://www.vikingelectronics.com/products/view_product.php?pid=217 - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 21, 2006 3:27 PM Subject: Re: [Asterisk-Users] asterisk + door opener FYI, astribanks all come with outputs that can be used for door openers, combined with this product from Vikingelectronics.com that plugs into any fxs port you should have a complete solution for a door: http://www.vikingelectronics.com/products/view_product.php?pid=99 They (viking) has a door opener that plugs into FXO as well (C2000-A) On 12/21/06, Thomas Kenyon [EMAIL PROTECTED] wrote: Jerry wrote: Hi Dovid, I am actually now working on massproducing door openers that will work with asterisk. It will have an rj45 port and then a port to plug the door opener in to. Please contact me off list if you are interested. This is an old message, but I was wondering if you are still doing this, and what the specs/cost are. Thanks, J. I'd be interested too, I was thinking of upgrading our door opener with a telephone line adapter and an FXO port from the linecard, but if I can do this without using an FXO port (and doesn't cost the earth) It would be great. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP problem
What zap device do you have that encodes/decodes g729? - Original Message - From: O.Kamal To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, December 18, 2006 4:37 PM Subject: Re: [asterisk-users] ZAP problem Why do I need g729 license?, i am not doing any transcoding in the middle. it is all g729 passthrough. softphone---asterisk---zap___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI History
- Original Message - On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote: What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Nothing wrong here. Can you possibly be a little more specific on why it isn't a problem? It's most likely how he is quitting the client. If you exit properly (exit or quit) it retains it but if you can cancel out (ctrl-c) it just drops. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disabling Features Temporarily
There is a company that I call that requires a * be dialed to break out of their IVR. The problem is Asterisk is grabbing that * for itself. Is there a way to get this sent? asterisk1*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# *2 Attended Transfer One Touch Monitor *1 Disconnect Call * *3 asterisk1*CLI show version Asterisk 1.2.12.1 built by root @ asterisk1.local on a i686 running Linux on 2006-09-27 19:41:58 UTC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dual Wan Router with Failover
We've never used it in a load balancing situation but it works great in a failover config. - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, November 14, 2006 10:41 AM Subject: RE: [asterisk-users] Dual Wan Router with Failover Sweet, now that is interesting http://hotbrick.com/produto.asp?tipo=2codPro=22 anyone have any comments on the load balancing capability of these? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dual Wan Router with Failover
http://hotbrick.com/ - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, November 14, 2006 9:38 AM Subject: RE: [asterisk-users] Dual Wan Router with Failover Are you looking for load balancing or failover. Also is there a cheaper way of implementing load balancing than $845 appliance? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 601 Phone can not find TFTP server
Are you sure you have the phone setup to use TFTP and not FTP? - Original Message - From: Klaverstyn, David C To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 02, 2006 1:51 AM Subject: [asterisk-users] Polycom 601 Phone can not find TFTP server Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying “Could not contact boot server, using existing configuration”. I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] plainvoip - down ???
- Original Message - From: J. Oquendo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 19, 2006 3:22 PM Subject: Re: [asterisk-users] plainvoip - down ??? Joseph wrote: Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails I can get there just fine. Your routes might be toasted [EMAIL PROTECTED] ~]# ping -c 10 plainvoip.com PING plainvoip.com (66.199.240.2) 56(84) bytes of data. ... --- plainvoip.com ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9013ms rtt min/avg/max/mdev = 75.531/78.550/80.349/1.418 ms, pipe 2 Depending on your location thought, there are issues with GBLX possibly due to a fiber cut either in VA or DC. No, the service is down. If you turn asterisk off, isn't your box still pingable? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] TDM2400P wiring.
- Original Message - From: C F [EMAIL PROTECTED] Sorry but that doesn't answer my question. My question is was: Which slot on the TDM2400 is Zap channel 1, which according to you will be Pair 1 (1 and 26) on the 66 block? Each card supports 4 lines so the card in Slot 6 would be ZAP/21-24 and Slot 5 would be ZAP/17-20 That would mean you need pairs 17-24 in your current config. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very high ping times from 7960 phones
- Original Message - From: Steve Glaus [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 22, 2006 4:25 PM Subject: [asterisk-users] Very high ping times from 7960 phones I've asked this here before and never really got a response, so I'll try again :) I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. Does anyone have any idea what might be causing this? I thought that it might just be a 'reporting' issue but there is definite latency there when I do an echo test. I'm running cisco sip firmware 8.2 on all the phones. All my Cisco phones show less than 75ms except for one (mine of course). I do have a switch in my cube that I use for extra ports and that's the only real difference. Do you have anything plugged into the extra network port on the phone? What's in between your phone and the asterisk server? _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very high ping times from 7960 phones
My asterisk server has 2 NICs . One with a public IP and one with an internal LAN IP. All the phones configure to the LAN IP so there's basically nothing between them. A 3com switch and that's it. basically nothing is wrong. I have a 3com switch in front of the one phone that reports the large time. Now I'm thinking it has something specifically to do with 3com switches and these phones. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] grandstream gxp 2000 does not display names whencalling out
- Original Message - From: Christopher Corn marco, can you explain what it is your recommending? i dial now from SIP phone to SIP phone. one being a grandstream gxp grandstream 100. but the gxp, when dialing, doesn't see the name of the person at the grandstream 100. i believe this should be picked up from the asterisk server, but im not sure. thanks. I was under the impression that the Grandstream 100 can not display anything other than the callerid number. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out
- Original Message - From: Christopher Corn To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, September 19, 2006 4:48 PM Subject: Re: [asterisk-users] grandstream gxp 2000 does not display nameswhencalling out im sorry if im not being clear. when im calling from my gxp to the grandstream 100.the gxp doesn't pullup the users name from grandstream 100. someone in another mail mentioned that this is not the way its supposted to work. my office does this, when i dial someones number, it displays their name. i wonder how this is done. thx. What phones/system do they use? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reload question
Whenever I issue a reload command I never seem to get any console feedback and the command never finishes. Is there a lock file or something I should check for? The system is still running and working just not reloading. Is there any way to find out where it's hanging? Connected to Asterisk 1.2.11 currently running on asterisk1 (pid = 31986) Verbosity is at least 10 asterisk1*CLI reload wait a few and then asterisk1*CLI reload The previous reload command didn't finish yet _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Source Directory of ASterisk
The source is not installed by default. If you use the trixbox repository you can use yum to install it or else: http://yum.trixbox.org/centos/4/SRPMS/asterisk-1.2.9.1-1.34876.src.rpm And in the future, you'll get more help if you post in the trixbox support forums instead of the Asterisk specific mailing list: http://www.trixbox.org/modules/newbb/ _Mobilcomhttp://www.mobilcom.net - Original Message - From: Tom Vile To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 28, 2006 12:49 PM Subject: Re: [asterisk-users] Source Directory of ASterisk /usr/src/asterisk On 7/28/06, Wasif [EMAIL PROTECTED] wrote: Hi,I am using TriBox 1.1.1/Asterisk. I want to know where I can find sourcedirectory of Asterisk in system so I can install Asterisk audio conversionmodule ( http://redice.krisk.org/res_conv-0.1.tgz) to convert ulaw promptsinto g729 prompts. It requires to point Asterisk source Include directory.Thanks___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd SIP timeout
I have two SIP providers that work fine during the day but it seems that during extended periods of non-use one provider gets stale and the refresh shows an odd number. A simple reload of SIP clears the problem but don't want to rely on cron as a solution. I qualify them both and I'm not behind NAT. Any ideas? asterisk1*CLI sip show registry HostUsername Refresh State plainvoip:5060 xxx70 105 Registered sip.provider.com:5060 724xxx-2147483 Registered asterisk1*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found asterisk1*CLI sip show registry HostUsername Refresh State plainvoip:5060 xxx70 105 Registered sip.provider.com:5060 724xxx 23 Registered asterisk1*CLI show version Asterisk 1.2.10 built by root @ asterisk1.local on a i686 running Linux on 2006-07-18 20:08:40 UTC _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP 8-3-0
Are you using the Non-CallManager version? _ Mobilcom http://www.mobilcom.net - Original Message - From: Tong [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 8:56 PM Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0 if you don't report it to cisco they won't know that bug exisit. - Original Message - From: Daryl Johnson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 4:05 PM Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0 Tim, I have seen the same 400 errors and the broken MWI... I backed up to 7.3... We'll see if Cisco corrects these in the next release... Daryl - Original Message - From: Tim Connolly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 12:06 PM Subject: [asterisk-users] Cisco 7960 SIP 8-3-0 Looks like the MWI broke on 8-3 also... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which application to open Zap channel?
This will just pick up the line exten = *01,1,Dial(ZAP/1/) _ Mobilcom http://www.mobilcom.net - Original Message - From: Carey O'Shea [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, June 14, 2006 9:48 AM Subject: [Asterisk-Users] Which application to open Zap channel? I'm sure this a very common and easy thing to do with Asterisk, but for the life of me I can't find the application that will allow me to open a Zap channel. Real world example: To be able to connect to an open Zap channel, so it would allow me to say, join in on a call that was originally answered by a PSTN phone (ie. just like you would by simply picking up another PSTN phone..!). There is ZapBarge, but allows no speaking, which is useless for this situation. Maybe I just have to use Dial in some way? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk receiving call from Panasonic TDA extension issue
Sorry someone screwing with permissions on my server bounced the 2 days worth of email after I posted this, any and all those lovely people who replied with suggestions from my post could you sent them again :-) James -Original Message- From: James Bean On Behalf Of Asterisk Mailing List Sent: Tuesday, 30 May 2006 12:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Asterisk receiving call from Panasonic TDA extension issue Asterisk, Zap and Libpri version from Asterisk SVN-branch-1.2-r27093 Error:- -- Accepting overlap call from '123' to '6' on channel 0/31, span 1 -- Starting simple switch on 'Zap/31-1' -- Hungup 'Zap/31-1' Primary Rate E1 30 trunks connecting between Asterisk and TDA200 Pansonic TDA200 has 1XX extensions Asterisk is setup with 6XX extensions If Asterisk calls a 1XX its not an issue, when 1XX calls Asterisk it looks like the phone system is dialing the digitals individually instead of at once so Asterisk is receiving the first 6 going I don't know 6 before it receives the rest of the digits from the TDA. Any clues as to if its possible to have asterisk wait for the rest of the digits, a wait of sorts, or I have to figure out how to make the TDA do it? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk receiving call from Panasonic TDA extension issue
Asterisk, Zap and Libpri version from Asterisk SVN-branch-1.2-r27093 Error:- -- Accepting overlap call from '123' to '6' on channel 0/31, span 1 -- Starting simple switch on 'Zap/31-1' -- Hungup 'Zap/31-1' Primary Rate E1 30 trunks connecting between Asterisk and TDA200 Pansonic TDA200 has 1XX extensions Asterisk is setup with 6XX extensions If Asterisk calls a 1XX its not an issue, when 1XX calls Asterisk it looks like the phone system is dialing the digitals individually instead of at once so Asterisk is receiving the first 6 going I don't know 6 before it receives the rest of the digits from the TDA. Any clues as to if its possible to have asterisk wait for the rest of the digits, a wait of sorts, or I have to figure out how to make the TDA do it? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7970 running SIP question
I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. It's so the phone routes the call to the correct server especially in a multiple server environment (ex: dialing a missed call) _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some ignorance here, what exactly is a Session Control Border? (Verizonis asking me about this)
It sounds like they referring to a load balancer that would provide one live external IP and handle sessions for multiple internal servers. _Mobilcomhttp://www.mobilcom.net - Original Message - From: Gabriel Afana To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, March 10, 2006 2:11 AM Subject: [Asterisk-Users] Some ignorance here,what exactly is a Session Control Border? (Verizonis asking me about this) Hey everyone, I am working with Verizon to terminate SIP calls for me. MY QUESTION: I notice the diagram shows only one asterisk server (softswitch). Since asterisk is not very good at clustering (appearing as one entity), I would have to have multiple servers (softswitches). You mentioned "You will give us the IP addresses of your softswitch(es) and that will allow communication between your network and ours." IfI gave you 3 IP addresses (to three asterisk softswitches), how would you hand off the calls unless there was some policy defining which calls go to which IPs. THEIR RESPONSE:I only represented one softswitch, because many of my previous customers only have one. Since you will have multiple softswitches, do you plan on implementing a session border controller? This would allow us to point the traffic to the SBC and thereby eliminating the worry of trying to separate the traffic. Are all the Asterisks going to be in the same location? S, what does she mean by a session border controller? As I understand, * is the SBC right? If not, how would an SBC fit into the picture with Asterisk? - Gabe P.S. And yes, I have thought about SER as a solution to this and have not yet given it a try. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7940/60 SIP 8.2
So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
- Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:42 AM Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2 So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. Sweet, guess I'll give it a go. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
I haven't been through everything line by line but I did notice a new Security Configuration where you can set an Encrypt Key _ Mobilcom http://www.mobilcom.net - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 11:20 AM Subject: Re: [Asterisk-Users] 7940/60 SIP 8.2 Yeah, this is the same procedure I went through with mine, worked like a charm, zero problems whatsoever... Anyone have any idea what if any the new features are of this firmware? Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
I believe they've done that the entire time. I've never known them to be real supportive of competing third party solutions. _ Mobilcom http://www.mobilcom.net - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 12:23 PM Subject: Re: [Asterisk-Users] 7940/60 SIP 8.2 Does that mean that since CCM supports SIP, Cisco will just make sure that their SIP images work with CCM? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970
tar zxfv *.cop - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 06, 2006 4:00 PM Subject: Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970 Ok, so, we've got the 7970 SIP Firmware now, but their readme is a little sparse... Anyone have any clue as to the upgrade procedure for a non-ccm5 system? (i.e. asterisk ;)) Aaron Julien Goodwin wrote: I've just recieved a copy of the new SIP firmware for the Cisco 7970, those of you with Cisco accounts may wish to try it (shock horror I'm sticking with SCCP). This coincides with the release of v8 firmware for all Cisco phones (and for those of you running Sergio's chan_sccp v8 works fine) The firmware is now also (and for the 7970 SIP, only) distributed in .cop files, these are actually just tarballs (.tar.gz) with a new name. The names are mangled, but relativly easy to figure out. Please note that I will not give this firmware out, nor point people to places where they may pirate it. Thanks, Julien ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970
It's just a tarball, extract it tar zxfv *.cop _ Mobilcom http://www.mobilcom.net - Original Message - From: Darren Wright [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 06, 2006 4:03 PM Subject: RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970 OK. I've got the COP SIP filehow do we use this thing on the 7970? -Darren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line Dropouts on E405P
')exten =
381,2,Record(newrecording.gsm)exten = 381,3,Festival('You
said')exten = 381,4,Playback(newrecording)exten =
381,5,Festival('Press 1 to continue or 2 to change your message')exten =
381,6,ResponseTimeout(5)
exten = t,1,Festival('Sorry, I did not get that')exten =
t,2,Goto(581,5)
exten = i,1,Festival('Sorry, that is an invalid choice')exten =
i,2,Goto(581,5)
exten = 1,1,System(/bin/mv /var/lib/asterisk/sounds/newrecording.gsm
/var/lib/asterisk/sounds/lm1/tempnew/${TIMESTAMP}.gsm)exten =
1,2,Festival('Thank you, your recording has been saved.')exten =
1,3,Festival('Press 3 to record another file or 4 to hang up')
exten = 2,1,Goto(581,1)exten = 3,1,Goto(581,1)exten =
4,1,Hangup
[parkedcalls]; Car movements emailed to PDexten =
388,1,SetMusicOnHold(random)exten = 388,2,VoiceMail,b280exten =
388,3,Playback(Goodbye)exten = 388,4,Hangup
exten = 390,1,playback(lm1/call_may_be_recorded)exten =
390,2,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/DNE|te405p-in,Zap/g4/211,1)
[emergency]exten = s,1,Dial(ZAP/g1/000)
exten = s,1,SetVar(SET_EMERG_FLAG=0)exten =
s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})exten =
s,n,SetGlobalVar(EMERGENCY=1)exten =
s,n,SetVar(SET_EMERG_FLAG=1)exten =
s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})exten =
s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)exten =
s,n,SoftHangup(${EMERGENCY_TRUNK}-1)exten = s,n,Wait(12)exten =
s,n,Goto(checkavail)exten = s,s+2(inprogress),Congestionexten =
s,checkavail+101(notavail),Goto(trunkbusy)exten =
h,1,GotoIf($[${SET_EMERG_FLAG} = 1]?3)exten =
h,3,SetGlobalVar(EMERGENCY=0)
[to-sip]
#include extensions_sip.conf
[queue_admin_ext]exten =
_2XX,1,Dial(ZAP/g4/${EXTEN},20,mA(lm_features/Queues/pls_say_thankyou_holding))exten
=
_3XX,1,Dial(SIP/${EXTEN},20,mA(lm_features/Queues/pls_say_thankyou_holding))exten
=
_7XX,1,Dial(ZAP/g4/${EXTEN},20,mA(lm_features/Queues/pls_say_thankyou_holding))
[lm1_functions]
#include extensions_lm1a.conf#include
extensions_js_play.conf#include
extensions_night_switch.conf___
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RE: [Asterisk-Users] Line Dropouts on E405P
wn extension (te405p-intelstra, 38165910, 2) exited non-zero on'Zap/29-1' -- Hungup 'Zap/29-1' -- Channel 0/27, span 1 got hangup request -- Channel 0/28, span 1 got hangup request -- Hungup 'Zap/43-1' == Spawn extension (te405p-intelstra, 38165909, 2) exited non-zero on'Zap/27-1' -- Hungup 'Zap/27-1'!! Got reject for frame 4, retransmitting frame 4 now, updating n_r!!! Got reject for frame 4, retransmitting frame 5 now, updating n_r!!! Got reject for frame 4, retransmitting frame 6 now, updating n_r!!! Got reject for frame 4, retransmitting frame 7 now, updating n_r!!! Got reject for frame 4, retransmitting frame 8 now, updating n_r!!! Got reject for frame 4, retransmitting frame 9 now, updating n_r!!! Got reject for frame 4, retransmitting frame 10 now, updating n_r!!! Got reject for frame 4, retransmitting frame 11 now, updating n_r! -- Channel 0/18, span 1 got hangup request -- Hungup 'Zap/40-1' == Spawn extension (te405p-intelstra, 38165908, 2) exited non-zero on'Zap/28-1' -- Hungup 'Zap/28-1' -- Hungup 'Zap/35-1' == Spawn extension (te405p-intelstra, 38165907, 2) exited non-zero on'Zap/18-1' -- Hungup 'Zap/18-1' -- Channel 0/26, span 1 got hangup request -- Hungup 'Zap/36-1' == Spawn extension (te405p-intelstra, 38165906, 2) exited non-zero on'Zap/26-1' -- Hungup 'Zap/26-1' -- Channel 0/31, span 1 got hangup request -- Channel 0/19, span 1 got hangup request -- Hungup 'Zap/32-1' == Spawn extension (te405p-intelstra, 38165903, 2) exited non-zero on'Zap/19-1' -- Hungup 'Zap/19-1' -- Channel 0/22, span 1 got hangup request -- Hungup 'Zap/37-1' == Spawn extension (te405p-intelstra, 38165905, 2) exited non-zero on'Zap/31-1' -- Hungup 'Zap/31-1'!! Got reject for frame 12, retransmitting frame 12 now, updating n_r!!! Got reject for frame 12, retransmitting frame 13 now, updating n_r!!! Got reject for frame 12, retransmitting frame 14 now, updating n_r!!! Got reject for frame 12, retransmitting frame 15 now, updating n_r!!! Got reject for frame 12, retransmitting frame 18 now, updating n_r!!! Got reject for frame 12, retransmitting frame 19 now, updating n_r!!! Got reject for frame 12, retransmitting frame 20 now, updating n_r!!! Got reject for frame 12, retransmitting frame 21 now, updating n_r! -- Hungup 'Zap/33-1' == Spawn extension (te405p-intelstra, 38165902, 2) exited non-zero on'Zap/22-1' -- Hungup 'Zap/22-1'!! Got reject for frame 16, retransmitting frame 16 now, updating n_r!!! Got reject for frame 16, retransmitting frame 17 now, updating n_r!!! Got reject for frame 16, retransmitting frame 18 now, updating n_r!!! Got reject for frame 16, retransmitting frame 19 now, updating n_r!!! Got reject for frame 16, retransmitting frame 20 now, updating n_r!!! Got reject for frame 16, retransmitting frame 21 now, updating n_r!!! Got reject for frame 16, retransmitting frame 22 now, updating n_r!!! Got reject for frame 16, retransmitting frame 23 now, updating n_r!!! Got reject for frame 18, retransmitting frame 18 now, updating n_r!!! Got reject for frame 18, retransmitting frame 19 now, updating n_r!!! Got reject for frame 18, retransmitting frame 20 now, updating n_r!!! Got reject for frame 18, retransmitting frame 21 now, updating n_r!!! Got reject for frame 18, retransmitting frame 22 now, updating n_r!!! Got reject for frame 18, retransmitting frame 23 now, updating n_r! -- Channel 0/17, span 1 got hangup request -- Hungup 'Zap/34-1' == Spawn extension (te405p-intelstra, 38165901, 2) exited non-zero on'Zap/17-1' -- Hungup 'Zap/17-1'!! Got reject for frame 21, retransmitting frame 21 now, updating n_r!!! Got reject for frame 21, retransmitting frame 22 now, updating n_r!!! Got reject for frame 21, retransmitting frame 23 now, updating n_r!!! Got reject for frame 21, retransmitting frame 24 now, updating n_r!asterisk1*CLI___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Dell blade servers
- Original Message - Sent: Friday, January 06, 2006 10:44 AM Subject: Re: [Asterisk-Users] Asterisk on Dell blade servers On Fri, Jan 06, 2006 at 02:17:47PM +, Bob Goddard said: On Friday 06 Jan 2006 08:11, Richard Scobie wrote: Supermicro do not do Opteron (or Athlon64) systems. Supermicro DO do Opteron. Model numbers please? Searching through SuperMicro's web site shows ZERO AMD based models. ONLY Intel. They do have a few chassis that claim to support AMD based motherboards, but NO superservers or motherboards. http://www.supermicro.com/Aplus/motherboard/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LinksysOne.com (New SIP phone, and more)
Another IP phone possibility for Asterisk. No, not the SPA941 (from the Linksys/Cisco/Sipura world)... Don't know much about it... but found this. Nothing on the datasheet says what it'll support really. http://newsroom.cisco.com/dlls/2005/eKits/Data_Sheet_IP_Manager_Phone.pdf But I found this that also talked about it being SIP based http://www.linksysinfo.org/modules.php?name=AvantGofile=printsid=438 http://www.linksysone.com Everything they want that isn't in the SPA941 ... PoE and integrated switch. Color screen. Price point $299 (estimated list price). Looks interesting. -- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647] ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] chan_bluetooth and Ericcson T68 problem
Hello Enky, We have encountered similar problems with various Ericsson Nokia phones. We couldn't get the channel driver to work 100%. However, we cannot actually tell whether it was our mistake or whether there was a problem with the channel driver. We tried to contact the driver's maintainer/creator but no luck... If you manage to find a solution for this problem we'd also be interested to know about it. Best regards, Vlasis. Enky wrote: Hi, I have read many pages and tried many things, but without any success. I have paired my ERICCSON T68 with the Asterisk PC. The Asterisk version is “Asterisk CVS-v1-0-11/19/05-14:52:52”. The chan_bluetooth is the last release, downloaded from “http://www.crazygreek.co.uk/data/pages/chan_bluetooth/latest.tar.gz”. It is all OK. I can dial from the Asterisk a number. The T68 dials it, but when the called party picks the phone and the call goes connected there is no any audio! Neither from or to the Asterisk. Here are a short logs: This is the initial log, when I start the Asterisk and it connects the T68. It seems OK: ---cut--- Asterisk Ready. *CLI Nov 19 15:15:45 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2041 try_connect: Initialised bluetooth link to device T68 [AG]T68 AT+BRSF=23 [AG]T68 ERROR [AG]T68 AT+CIND=? [AG]T68 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG]T68 OK [AG]T68 AT+CIND? Nov 19 15:15:46 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:417 set_cind: Audio Gateway T68 got signal [AG]T68 +CIND: 5,5,0,1,1,0,0,0,0,0 [AG]T68 OK [AG]T68 AT+CMER=3,0,0,1 [AG]T68 OK [AG]T68 AT+CLIP=1 [AG]T68 OK [AG]T68 AT+CGMI=? [AG]T68 OK [AG]T68 AT+CGMI [AG]T68 ERICSSON [AG]T68 OK ---cut--- This is when I dial a number. It seems OK too, but no audio when connects: ---cut--- -- Executing Dial(SIP/222-3885, BLT/T68/123|60) in new stack [AG]T68 ATD123; -- Called T68 [AG]T68 OK [AG]T68 +CIEV: 8,1 -- BLT/T68 answered SIP/222-3885 [AG]T68 +CIEV: 2,4 [AG]T68 +CIEV: 2,5 ---cut--- And this is when I interrupt the dialed call: ---cut--- [AG]T68 AT+CHUP == Spawn extension (default, 2002, 1) exited non-zero on 'SIP/222-3885' [AG]T68 OK Nov 19 15:18:06 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2493 rd_close: Device T68 disconnected, scheduled reconnect in 5 seconds: Connection reset by peer (errno 104) Nov 19 15:18:11 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2041 try_connect: Initialised bluetooth link to device T68 [AG]T68 AT+BRSF=23 [AG]T68 ERROR [AG]T68 AT+CIND=? [AG]T68 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG]T68 OK [AG]T68 AT+CIND? [AG]T68 +CIND: 5,5,0,1,1,0,0,0,0,0 [AG]T68 OK [AG]T68 AT+CMER=3,0,0,1 [AG]T68 OK [AG]T68 AT+CLIP=1 [AG]T68 OK [AG]T68 AT+CGMI=? [AG]T68 OK [AG]T68 AT+CGMI [AG]T68 ERICSSON [AG]T68 OK ---cut--- Please someone to help me :) Thank you in advance! ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] h323 question
Angelito Manansala wrote: yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! Hello, As far as I know Asterisk cannot disentangle RTP from signaling in either SIP or H323 at least until now. I'd also be interested to know if this option is available now in case I've missed something... Best regards, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Eicon Diva Server query
Avi Miller wrote: Hello gurus! I've given up on crappy passive ISDN cards and am heading into the wild world of real, Active Super Dooper Server boards. I have a choice of two Eicon Diva Server cards: Eicon Diva Server 4BRI Eicon Diva Server V-4BRI Hello, We've been using an Eicon Diva Server 4BRI with a RH 9 installation (kernel 2.4.20-8). It works great in both TE and NT mode. I assume that it will work equally great with a 2.6 kernel... Best regard, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] unexpected debug output from console
- Original Message - From: Jason Pyeron [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, November 15, 2005 11:09 AM Subject: [Asterisk-Users] unexpected debug output from console Why am I getting debug from server--sip.broadvoice.com on the console when debug and verbose are off? And can it be fixed? The server is 192.168.1.10, my laptop with softphone is 192.168.1.103 testserver*CLI set verbose 0 testserver*CLI sip no debug You didn't seem to get a response when you set verbose ex: voip*CLI set verbose 0 Verbosity is now OFF voip*CLI _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)
If anyone is interested I'm (slowly) developing a GPL'd Java applet that works as an IAX softphone. I should have a test version out at the end of the week for a limited number of testers. Tim. http://www.westhawk.co.uk/ Hello Tim, We'd be interested to test the client... Best regards, Vlasis Hatzistavrou Technical Director CEO Kinetix Tele.com Hellas Ltd. Monastiriou 9 Enotikon 546 27 Thessaloniki Greece Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: [EMAIL PROTECTED] http://www.kinetix.gr ___ 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] spandsp - fax is just blank pages
pbo 808 wrote: I've done quite a bit of googling and haven't found a solution to my problem. I've got the Digium dev kit (wctdm11b) set up and working. I've compiled spandsp and can receieve faxes from eFax (www.efax.com) but the pages are blank. The page count is correct, in that if I fax a two page document, my tiff file has two pages, but they are white blank pages. I found one similar post here http://lists.digium.com/pipermail/asterisk-users/2005-April/103069.html, but haven't seen a solution. Any ideas? ___ Hello, I have noticed the same problem in my tests with spandsp. I think it has to do with the format of the tiff file, but I couldn't find the reason... I hope that someone in this list who has solved this problem can share the solution with us. Best regards, Vlasis Hatzistavrou. ___ 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: TE100P connecting to non PRI, ISDN interfaces
Hello, i've googled and can't find a definite answer, so here goes: I have purchased the Digium TE100P, and am setting up the connection, however the telco i'm supposed to work with does not support PRI/ISDN E1 connections. They only support E1/R2 lines. Is there a way i can make the TE100P work with this? I've not seen any zaptel.conf that supports this. Any workarounds? Thanks for any help! Ian ___ 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] Revision I Board TDM04b
need latest zaptel source _Mobilcomhttp://www.mobilcom.net - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, June 28, 2005 2:53 PM Subject: [Asterisk-Users] Revision I Board TDM04b I cannot get this thing to work. Anyone know of any tricks? ___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] HELP Cisco - can't find P0S3-07-4-00.sbn
OS79XX.TXT should contain: P003-07-4-00 _ Mobilcom http://www.mobilcom.net - Original Message - From: Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 31, 2005 11:59 PM Subject: [Asterisk-Users] HELP Cisco - can't find P0S3-07-4-00.sbn I have a problem, I'm working with firmware SIP 6.3 installed on my Cisco phone and works fine, and I have the 7.4 firmware version to upgrade: [EMAIL PROTECTED]/home/alex/central/P0S3-07-4-00 ls -l total 2.3M -rw-r--r--1 root root 126K mar 10 15:33 P003-07-4-00.bin -rw-r--r--1 root root 578K mar 10 15:44 P0S3-07-4-00.bin -rw-r--r--1 root root 461 mar 10 16:01 P0S3-07-4-00.loads -rw-r--r--1 root root 579K mar 10 15:45 P0S3-07-4-00.sb2 -rw-r--r--1 root root 127K mar 10 15:33 P003-07-4-00.sbn -rw-r--r--1 root root 15 mar 10 15:33 OS79XX.TXT -rw-r--r--1 root root 895K abr 13 23:30 P0S3-07-4-00.zip When I try to upgrade to the 7.4 firmware I get this log: uploading OS79XX.TXT uploading P0S3-07-4-00.bin uploading P0S3-07-4-00.loads uploading P0S3-07-4-00.sb2 can't find P0S3-07-4-00.sbn - Aborted My phone is asking for a P0S3-07-4-00.sbn file, and can't find it in the Cisco distro. Perhaps a Cisco bug? Any idea? Regards, -- Ing CIP Alejandro Celi Mariátegui [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 ___ 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] new cisco ip video phone?
Any chance it's the phone mentioned here? http://voxilla.com/voxstory134.html _ Mobilcom http://www.mobilcom.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lex Lethol Sent: Thursday, May 26, 2005 2:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] new cisco ip video phone? Hi, Just finished watching the season finale of '24' the TV series. Throughout the series they have been showcasing Cisco hardware especially Cisco IP phones (7970's). On the last episode or two they showed what seemed to me a new cisco IP video phone. It stands just as a 12 lcd screen with the cisco branding/logo and letters just as the 79xx series. I wonder if this is a new cisco model thats ready to roll out. It looks great, but then again, I doubt they will support SIP on it (at least on release) Anyone else know anything on this? Lethol ___ 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 delivers CID even when restricted?
I noticed the same problem back when they fixed their problems. I open a ticket (CAS-23281) on 5/7 and as far as I can tell, it's still open. I have my outbound CID blocked from the their webpage but it shows a number when I place calls. To make it even more a problem, the number it shows (202-556-) is not my number! _ Mobilcom http://www.mobilcom.net - Original Message - From: Johnathan Corgan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 2:19 AM Subject: [Asterisk-Users] Broadvoice delivers CID even when restricted? I can call my Broadvoice DID from a outbound caller-id blocked phone, and BV happily delivers the CID to Asterisk (and then on to my IP phone display.) I've tested with the *67 prefix from a PSTN phone to make sure it was supposed to be blocked. The number is always correct, but sometimes the the caller ID name is set to something funky (like a CO or switch center name.) I *think* this started happening after they came up from the meltdown a couple weeks ago. Is caller ID blocking implemented by sending the cid information anyway, but with a bit that says don't give to end user? I guess BV would be ignoring this bit. Anyone else experience this with BV and Asterisk? -Johnathan ___ 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] Cellsocket help needed
I have a cellsocket working with a Nokia 6150 right now. Funny the
model I bought was a cellsocket for a Nokia 5110, and for some reason,
it wont work with the 5110 unit i put in. The 6150 works like a
charm, though. For some reason, it ignores the first two characters
of the phone number you dial, so i had to do something like:
PAUSE=**
ZERO=0
SHARP=#
;
; Outbound to 5nxx- goes via: CellSocket
exten = _59X,1,Dial(Zap/1/${PAUSE}${ZERO}${EXTEN:1}${SHARP})
exten = _59X,2,Congestion
The ZERO is used to call local long distance, and the SHARP is needed
by cellsocket
to tell it that it ends the number stream.
On 5/9/05, Manny A. Wise [EMAIL PROTECTED] wrote:
I need help from someone who has a working cellsocket, I have received
couple email of people who wanted to help, but they just think they know how
it supposed to work, but they don't have a working units, and they confused
more..I need someone with a working solution to get my cellsocket going.
Thanks!!!
Write offlits @ mawise (AT) mail.com
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Re: [Asterisk-Users] Broadvoice Issues
- Original Message - From: Rich Adamson [EMAIL PROTECTED] Until BV as a company steps up to the plate, it serves no useful purpose to bitch at them. Their doing exactly what they said they would do... nothing. But it's not really about what the end user is actually running if nobody can authenticate or pass any calls in or out. The only real difference is they have a built-in excuse for Asterisk users to not support their problems. I just had to switch off of proxy.dca because of not being able to make any outbound calls. Once I was on proxy.mia everything works fine so I was able to call 611 and talk to support. He said they new of a problem and was told it would be fixed in 3 hours (3:30 pm EST). Although I'm not sure if it was 3 hours from when I asked or when he was told. So they know they have a problem but don't like announcing it anywhere. Does anybody know if VoIP carriers/providers are held to the same standard for FCC outage reporting as other service providers? ___ Mobilcom http://www.mobilcom.net ___ 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] IP Phones for home use?
- Original Message - From: Neil Cherry [EMAIL PROTECTED] What are your recommendations for a slightly fancy home phone? Cisco 7940 Great phone. Great feel of quality (solid). And you can put a logo on it. ___ Mobilcom http://www.mobilcom.net ___ 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] xpro codecs and asterisk
Please show your dialing context from extensions.conf _ Mobilcom http://www.mobilcom.net - Original Message - From: Dov Bigio To: asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 11:01 AM Subject: [Asterisk-Users] xpro codecs and asterisk Hi all, I am trying to make a call from an X-Pro with only the G.729 codec enabled to another with both G.711 and g.729. The Asterisk version is 1.0.3 and canreeinvite is set to Yes. What happens is that I got an 403 - Forbidden response and setting verbose 10 in Asterisk I can see the message: May 2 15:47:36 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/victor-a02d(4) to SIP/dediana-1fd9(256) -- SIP/victor-a02d is ringing -- SIP/victor-a02d answered SIP/dediana-1fd9 May 2 15:47:38 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/dediana-1fd9(256) to SIP/victor-a02d(4) May 2 15:47:38 WARNING[6690]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make SIP/dediana-1fd9 compatible with SIP/victor-a02d If I do the same when both softphones have only G.711 set, everything works fine. It seems that Asterisk tries to use the first codec in SDP and ignores others. Does it make sense? Thanks in advance. Dov ___ 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] Is there any chance to bring Skype andAsteriskUser together?
I think the proper solution would be to use the proprietary Skype API for Linux and create an asterisk extension for it. There is a $1050 bounty on voip-info.org[1] but i don't think there are any takers for it yet. :( Another suggestion was ... to either get a Skype compatible ATA or FXS/FXO adapter, and just live with that, that's probably the closest you'll get to it any time soon [1] http://www.voip-info.org/wiki-bounty+skype On 5/4/05, Dean Collins [EMAIL PROTECTED] wrote: You could run an automated session out your speaker/mic to an incoming fxs circuit but to answer your question - No. Never heard it happen before. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili Sent: Tuesday, May 03, 2005 6:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Is there any chance to bring Skype andAsteriskUser together? What do you mean? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki Sent: Tuesday, May 03, 2005 3:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Is there any chance to bring Skype and AsteriskUser together? Hi, is there any chance to bring Skype and Asterisk User together? Regards, Kib ___ 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 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 limits???
- Original Message - From: Tim Connolly To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; 'Kerry Garrison' Sent: Sunday, May 01, 2005 2:50 AM Subject: RE: [Asterisk-Users] Broadvoice limits??? Broadvoice. Seems to be no limit on inbound, but I found any channels after 5 outbounds would get an immediate disco. Guess I'll have to stick to Vonage to blast into the local radio shows. Or maybe 5 on BV, 5 on Vonage, and X on the PRI. - Are you manually dialing out that many times or have you got some script to do it for you? Would be nice if there was a *66 feature (Automatic Callback Activation). ___ Mobilcom http://www.mobilcom.net ___ 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] voip connection problems
http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-04-187A1.pdf http://www.wi-fiplanet.com/voip/article.php/3390671 http://www.cybertelecom.org/voip/Fcc.htm (scroll down) and of course: FCC To Require 911 for VoIP http://www.newsfactor.com/story.xhtml?story_id=33733 - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 1:54 AM Subject: Re: [Asterisk-Users] voip connection problems trixter http://www.0xdecafbad.com wrote: a couple weeks ago the FCC (america) ruled that all voip providers that connect to the PSTN (vonage, broadvoice, voicepulse, etc) have to have CALEA support (wiretap equipment for law enforcement). Failure to comply is a $10,000 fine per day. Could you please provide a reference for this assertion? Thx. B. ___ 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] Traffic Testing
SIPp is a free Open Source test tool / traffic generator for the SIP protocol http://sipp.sourceforge.net/ On Fri, 2005-04-29 at 14:14 +0200, Nils Ohlmeier wrote: The homepage http://sipsak.org contains some examples. If you need help with special cases drop me a line. Regards Nils Ohlmeier On Friday 29 April 2005 02:54, Anton Krall wrote: Can you send some command line examples on how to use it? Thx! |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |[EMAIL PROTECTED] |Sent: Jueves, 28 de Abril de 2005 07:05 p.m. |To: asterisk-users@lists.digium.com |Subject: RE: [Asterisk-Users] Traffic Testing | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] Behalf Of Anton | Krall | Sent: Thursday, April 28, 2005 6:07 PM | To: 'Asterisk Users Mailing List - Non-Commercial Discussion' | Subject: [Asterisk-Users] Traffic Testing | | | Guys, is there any way to generate simulated traffic via sip or IAX2 | for testing cpu load and asterisk? (sip client simulation, etc)? | |yes, use sipsak utility | |-- |___ |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 -- René Mayorga Internet Data El Salvador Telecom S.A. de S.V. Tel:(503) 247-7246 (503) 247-7156 Cel:(503) 962-8205 ___ 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 Down?
If I dial into my number I get nothing but dead air and then it hangs up. the really odd thing is the call makes it to my sip phone but it's just dead air if I answer. The bad thing for me is my outbound does not work as everything times out. Only call I can reliably place is to their support number but no matter which option I pick I get sent right to a busy signal. This is all so lovely... - Original Message - From: Max Clark [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 25, 2005 1:08 PM Subject: [Asterisk-Users] Broadvoice Down? Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- Max Clark max [at] clarksys.com http://www.clarksys.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 ___ 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] sangoma S508/FT1 ISA
did you try [EMAIL PROTECTED] f. On Sat, 2 Apr 2005, Michael Bielicki wrote: it wan't do channelised stuff so it want be of any use for voice On Apr 1, 2005 2:46 AM, Neal Walton [EMAIL PROTECTED] wrote: Hi, Does anyone have any experience with the Sangoma S508/FT1? I can't seem to find very much information on it, and Sangoma has not responded to my e-mail. The Sangoma wanpipe driver doesn't seem to support TDM on this card, but I feel certain that the card can handle it. I hope someone knows how to make this card work so I don't have to hack the firmware. Regards, Neal ___ 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 -- Michal Bielicki http://www.asterisk.com.pl/ ___ 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] Implant GIPS's codec to Asterisk
Speaking to this, does anybody know anyone who will join me in harassing Grandstream in implementing speex support in their phones? f. On Wed, 30 Mar 2005, Steve Kann wrote: Steve Underwood wrote: Gustavo GarcĂa wrote: Hi everybody, GIPS have different products, not only codecs: * Voice enhancements: packet loss concealment algorithms, noise concealment, jitter buffer, agc, aec (can be used with any codec) * Codecs: iLbc (free), ISAC, G711 Wideband... You can include in asterisk voice enhancements and use them iLBC for example, for increasing the quality mainly in face of the packet loss, without using wideband codecs. I'm not a GIPS employee :-), you can view more information in the GIPS website. G. From what I have seen it appears those GIPS products are not particularly sophisticated. For example, have you any reason to believe they can achieve better jitter and packet loss handling than * with the new jitter buffer and PLC? That is not the world's most sophisticated, but as far as I get tell it is about on par with the GIPS offering. Does anyone have any evidence to the contrary? I've read about GIPS' jitterbuffer stuff, and I think that our jitterbuffer implementation offers basically the same featureset. I would imagine that at this point, GIPS' implementation is probably better tested, but would be much more difficult to integrate into *. As far as the other DSP functions you mention, libspeex provides all of these, in varying degrees of progress (i.e. AGC, VAD, Denoise work pretty well, AEC does not yet work very well). Also, as far as wideband codec support, Speex supports both wideband (16khz) and ultra-wideband (32khz) modes, and these both work really well, as I use them in other applications. The work to include these (free, as in speech and beer) codecs would probably be roughly the same as for the wideband iLBC (not free, as in speech _or_ beer), and would benefit everyone out-of-the box, as opposed to just those who want to go through the trouble (and expense) of licensing a commercial codec. -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 ___ 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] Looking for SS7 design input
How about simply doing a Q.931 to SIGTRAN conversion module would that not be simpler to implement? -=Francois=- On Wed, 30 Mar 2005, NVC List Manager wrote: On Wednesday 30 March 2005 11:16, TC wrote: I am looking for input on what an SS7 interface to Asterisk should look like and what it will need to be of any use. If you don't want to help then don't whine and complain about how you don't need SS7. All comments made in jest are welcome; points will be awarded for cheekiness and good puns. The code won't be written for a while because the design must predate the coding. But please let me know if you would like it done a certain way or need a certain feature. CLASS 5 or 4 SCP, SSP, SCT Local Exchange MU2A, MU3A SG Maybe you could throw some effort over here http://ss7box.com/asterisk.html This design to me looks well thought out, scaleble, GPL :) Hmm, my understanding is that Mike is developing a commercial SS7. -- NVC List Manager (Not Asterisk's) ___ 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] Use of asterisk to make use of IP phone speakerphones as a baby monitor....
Is this possible? f. ___ 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] * Mobile Phone Mobile Network
Another option is something like cellsocket http://www.cellsocket.com I haven't tried these, but some positive experiences posted in some sites ive been googling seem encouraging. There are models for motorola and nokia phones. On Mon, 21 Feb 2005 15:21:59 +1100, Mathew McKernan [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Uzzell Sent: Monday, 21 February 2005 2:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] * Mobile Phone Mobile Network Ok I have a question. Seen it come and go around the mailling list for a while but never really seen an answer that seems to sort it out. What is needed is some interface from * Mobile Phone Mobile Network Service. At this point all the providers in AUS that I have found are charging a Premium Rate for Land Line Mobile Network services. What I would like to do is be able to purchase a low rate Mobile SIM that I can chuck into a Mobile Phone and have it setup so that I route the Mobile calls through it. Rembering that most if not all mobile phones can be accessed via RS232 interface. Anyone done this or seen it done or know how to do it using * and whatever? Cheers David ___ 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 Hi David, Have a look at some second hand Ericson kits on Ebay. They had special units, that basically had a normal GSM Ericson phone in them. But on the side had a normal Australian 610 socket and rj11 socket. You could simply interface this into your digium cards as a normal pstn line. They were originally designed for the exact purpose you want for coupling with existing telephone systems. They are also used for connection to fire signalling units and alarm systems. Thanks Mathew McKernan Digital World Computers Maribyrnong VIC ___ 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 digit dropping
I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option related to DTMF). Anyone else getting similar drops? Any solutions? Is http://connect.voicepulse.com/ , using IAX, any better? ___ 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 digit dropping
I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option related to DTMF). Anyone else getting similar drops? Any solutions. Is http://connect.voicepulse.com/ , using IAX, any better? ___ 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] Jeff Pulver quoted talking about Asterisk...
VOIP pioneer predicts a roiling 2005 for IP telephony Eetasia.com (subscription) - USA Open source software communications will begin to influence the VoIP market in a big way next year, according to VoIP pioneer Jeff Pulver. ... http://www.eetasia.com/article_content.php3?article_id=8800354924 (use BugMeNot) or ... View it at http://lenny.tropiano.org/voip-2005.pdf ___ 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] Zaptel HDLC (NetHDLC) errors on modprobe, Linux 2.6 kernel
I have my Linux 2.6 kernel with the necessary HDLC config and also recompiled zaptel accordingly. On modprobe, I get: Found a Wildcard: Digium Wildcard T100P T1/PRI Debug: sleeping function called from invalid context at mm/slab.c:2000 in_atomic():0[expected: 0], irqs_disabled():1 [0211e605] __might_sleep+0x82/0x8c [02144643] kmem_cache_alloc+0x1d/0x57 [42ad7a2c] zt_ctl_ioctl+0x112c/0x16b0 [zaptel] [022d6c00] __cond_resched+0x14/0x39 [428a315a] ext3_get_inode_loc+0x4f/0x210 [ext3] [0217269c] d_instantiate+0xa3/0xa9 [021729ea] d_splice_alias+0x145/0x14e [428a4ec9] ext3_lookup+0x70/0x89 [ext3] [0216802f] real_lookup+0x6e/0xd2 [02171457] dput+0x1b/0x287 [0216901d] link_path_walk+0xd3c/0xdf7 [02163ffc] cdev_get+0x33/0x68 [02163f7a] exact_lock+0x7/0x11 [0221850d] kobj_lookup+0x132/0x194 [02163f70] exact_match+0x0/0x3 [0216d078] sys_ioctl+0x23d/0x2a0 divert: not allocating divert_blk for non-ethernet device hdlc0 Registered tone zone 0 (United States / North America) Using ESF/B8ZS coding/framing Calling startup (flags is 4099) Using ESF/B8ZS coding/framing Calling startup (flags is 4099) divert: no divert_blk to free, hdlc0 not ethernet Debug: sleeping function called from invalid context at mm/slab.c:2000 in_atomic():0[expected: 0], irqs_disabled():1 [0211e605] __might_sleep+0x82/0x8c [02144643] kmem_cache_alloc+0x1d/0x57 [42ad7a2c] zt_ctl_ioctl+0x112c/0x16b0 [zaptel] [02158793] rw_vm+0x2df/0x331 [02171457] dput+0x1b/0x287 [0216901d] link_path_walk+0xd3c/0xdf7 [0216480a] cp_new_stat64+0xee/0x10d [428abfb6] ext3_permission+0x0/0x153 [ext3] [02163ffc] cdev_get+0x33/0x68 [0216d078] sys_ioctl+0x23d/0x2a0 Zaptel config /etc/zaptel.conf: span=1,1,0,esf,b8zs nethdlc=1-24 loadzone = us defaultzone=us The HDLC config does work, I am able to sethdlc and bring up the connection not sure why we're getting those errors above, is there anything that can be done to make it cleanly load? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SysMaster and GPL Violation
Do you know what hardware is Sysmaster using as a TDM interface? Is it relying on Asterisk to perform host based processing? If so, I would not be surprised that if they are OEMming lots of Digium boards, there is no wonder here why Digium keeps it cool, if not, well... -=Francois=- [EMAIL PROTECTED] On Fri, 12 Nov 2004, Jeremy McNamara wrote: Brian West wrote: So all you Sysmaster owners run strings on the 'voipgw' binary that runs on those boxes and you'll see that its asterisk. If you have doubts I'll post more proof. I have had a few customers inquire about sysmaster's usage of Asterisk. One of them granted me access to his system, to determine what was under the hood. All I had to do was crack open the computer, pull the hard drive, jack it into my own Linux box and mount it. After a very quick scan of the hd, which is a pretty typical, yet minimal, Linux OS, I found the 'voipgw' application. Here is a selected portion of the strings output: [EMAIL PROTECTED] ~]# strings voipgw | grep Mark Written by Mark Spencer [EMAIL PROTECTED] [EMAIL PROTECTED] ~]#strings voipgw | grep CVS Asterisk CVS-05/30/03-20:39:27 built by [EMAIL PROTECTED] on a i686 running Linux CVS-05/30/03-20:39:27 Asterisk CVS-05/30/03-20:39:27, Copyright (C) 2000-2002, Digium. Asterisk CVS-05/30/03-20:39:27, Copyright (C) 1999-2001 Linux Support Services, Inc. [EMAIL PROTECTED] ~]# strings voipgw | grep Asterisk Asterisk IO Dump: %d entries, %d max entries Asterisk Schedule Dump (%d in Q, %d Total, %d Cache) Started Asterisk Event Logger Asterisk Event Logger Started Restarted Asterisk Event Logger Asterisk Event Logger restarted Asterisk Dynamic Loader Starting: -= Registered Asterisk Alternative Switches =- -= Registered Asterisk Applications =- Asterisk PBX Core Initializing Asterisk CVS-05/30/03-20:39:27 built by [EMAIL PROTECTED] on a i686 running Linux Asterisk %s cancelled. Asterisk %s ending (%d). Asterisk ending (%d). Preparing for Asterisk restart... Restarting Asterisk NOW... Exit Asterisk Shut down Asterisk imediately Gracefully shut down Asterisk Restart Asterisk immediately Restart Asterisk gracefully Restart Asterisk at empty call volume Disconnected from Asterisk server Connected to Asterisk %s currently running on %s (pid = %d) Asterisk CVS-05/30/03-20:39:27, Copyright (C) 2000-2002, Digium. -r Connect to Asterisk on this machine Asterisk CVS-05/30/03-20:39:27, Copyright (C) 1999-2001 Linux Support Services, Inc. Asterisk already running on %s. Use 'asterisk -r' to connect. Asterisk Ready. Asterisk Asterisk Console on '%s' (pid %d) Loads the specified module into Asterisk. Unloads the specified module from Asterisk. The -f Shows Asterisk modules currently in use, and usage statistics. Shows Asterisk version information. Exits Asterisk. Causes Asterisk to abort an executing shutdown or restart, and resume normal Shuts down a running Asterisk immediately, hanging up all active calls . Causes Asterisk to not accept new calls, and exit when all Causes Asterisk to hangup all calls and exec() itself performing a cold. Causes Asterisk to stop accepting new calls and exec() itself performing a cold. Causes Asterisk to perform a cold restart when all active calls have ended. [EMAIL PROTECTED] ~]# strings voipgw | grep ast_ ast_restore_tty ast_default_amaflags ast_pbx_outgoing_app ast_translate ast_io_add ast_sendtext ast_closestream ast_set_indication_country ast_cdr_start ast_dsp_digitreset ast_context_remove_switch ast_context_add_include ast_sched_add_timer_func ast_verbose ast_async_goto ast_indicate ast_channel_register [snipped for posting] A complete strings output is here: http://www.nufone.net/downloads/voipgw.txt Upon confronting sysmaster with this fact, they (Mike Fahey, Ray Martinez, and other more technical people) completely denied their usage of Asterisk insisting they developed their solution in-house. I too demand sysmaster either pay Digium for a non-gpl license or publicly admit the fact that they have repackaged Asterisk and contribute enhancements to Asterisk back to the GPL. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] * and Verisign SIP-7 service
So, I think that Asterisk will provide the functionality that you desire. However, I don't know if SIP-MGCP calls can presently be completed without Asterisk proxying the media stream, so you may have performance issues. Perhaps someone else can address that. In what context will Asterisk will require proxying the media stream? I have a simple setup whereby I make my FWD account ring my Mediatrix 2102 as an extension to my Asterisk and the delay is horrific f. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] * and Verisign SIP-7 service
On Thu, 21 Oct 2004, Kevin P. Fleming wrote: No, Asterisk cannot control an MGCP gateway at this time. If the AS5400 is in MGCP mode, it will be expecting a softswitch to control it, and it will Can you expand ... I'm hoping to use Asterisk to do SIP to MGCP mediation ... what does it not work? -=Francois=- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: GSM to g729 Conversion
and the dll runs under wine? f. On Tue, 19 Oct 2004, Pavel Jezek wrote: you can use vovida's open g729 sample code, look to: http://www.voiceage.com/codecsite/openinit_g729.php PJ - Original Message - From: Matthew Boehm Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Tuesday, October 19, 2004 6:03 PM Subject: Re: GSM to g729 Conversion There is no way to convert existing files to g729? The only reason we need the licenses is to access voicemail since they are in GSM. All our phones have g729 built in. But if you try and access VM, you get that No coversion for GSM to g729 error. But if all the voicemail sounds where in g729, then we don't need the licenses. Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer caller but on no answer, return to transferee...
So Asterisk gurus out there, is there a nice clean way in the dialplan to determine if the caller is coming from a transferred call, and on the unavailable context in the dial, instead of going to e-mail go back to the transferee? If anyone has this sort of logic or could spit out an extensions.conf snippet; I'd be grateful. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some photos from Astricon 2004
These taken tonight (9/22/2004) at the Expo and Reception Enjoy. http://photos.tropiano.org/gallery/astricon-2004 Lenny ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Astricon
Does anyone know if the Marriott has Wi-Fi? LAN connection in the room? According to the STSN (www.stsn.com) hotel locator, the Marriott does have in room wired access. Wireless access and Meeting room access. At $9.99/day (cheaper usually if you buy blocks of in multiple days) locked to a MAC address a NAT router would help multiple computers to share... Usually you can select a private or public (no firewall) address when signing up... I suspect the in-meeting room access might be free if worked out from the hotel. I see others are bringing their Linksys WRT54GS routers that'll be great. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Caller-ID name lookup via anywho.com
Did I see something on here about using an AGI script to do reverse lookups via anywho.com? I have a PRI that only gets caller-id number and no Alpha. [...] I put a copy of it here... http://www.voiping.com/calleridnamelookup.agi It was written by James Golovich [EMAIL PROTECTED] and requires the Asterisk::AGI perl bindings, but works... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePulse Connect DTMF with IAX2
Is there anyone out there who has VoicePulse Connect working with DTMF?
I've been unable to get it to work from the start, and the recent
VoicePulse updates
did not help.
A caller to my DID's hears Asterisk, but pressing DTMF does nothing:
On call setup iax2 debug shows:
-
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK Timestamp: 2ms SCall: 3 DCall: 00037
[66.234.228.144:4569]
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW Timestamp: 1ms SCall: 00037 DCall: 0
[66.234.228.144:4569]
VERSION : 2
CALLED NUMBER : 5107400469
CALLING NUMBER : 5105408421
CALLING NAME: 5105408421
LANGUAGE: en
CALLED CONTEXT : INGRESS
USERNAME: germanium
FORMAT : 4
CAPABILITY : 4
ADSICPE : 2
DATE TIME : 152969415
CLI iax2 show registry
Host UsernamePerceived Refresh State
66.234.228.170:4569 X 208.184.214.241:4569 60 Registered
---
My config is:
iax.conf:[general]
iax.conf:disallow=all
iax.conf:allow=ulaw
iax.conf:allow=ilbc
iax.conf:allow=gsm
iax.conf:allow=adpcm
iax.conf:allow=alaw
iax.conf:jitterbuffer=no
iax.conf:delayreject=no
iax.conf:register = :[EMAIL PROTECTED]
iax.conf:
iax.conf:[voicepulse-in-01]
iax.conf:type=user
iax.conf:context=voicepulse-test
iax.conf:auth=rsa
iax.conf:inkeys=voicepulse01
extensions.conf:[general]
extensions.conf:static=yes
extensions.conf:writeprotect=yes
extensions.conf:
extensions.conf:[globals]
extensions.conf:[default]
extensions.conf:[voicepulse-test]
extensions.conf:exten = _NXXNXX,1,Playback(beep)
extensions.conf:exten = _NXXNXX,2,SayDigits(${EXTEN})
extensions.conf:exten = _NXXNXX,3,Goto(testdtmf|s|1)
extensions.conf:
extensions.conf:[testdtmf]
extensions.conf:exten = s,1,Background(beep)
extensions.conf:exten = s,2,ResponseTimeout(60)
extensions.conf:exten = _x,1,SayDigits(${EXTEN})
extensions.conf:exten = _x,2,Goto(testdtmf|s|1)
extensions.conf:exten = i,1,Goto(testdtmf|s|1)
extensions.conf:exten = t,1,Hangup
I'm running:
Asterisk CVS-HEAD-08/01/04-22:51:56, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-HEAD-08/01/04-22:51:56 currently running on
skip (pid = 28611)
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[Asterisk-Users] How to minimally configure modules.conf loading?
I'm trying to somewhat reduce the security risk of Asterisk, by loading less modules. In my installation I use SIP and IAX2 for incoming calls, and that's it. No voicemail, no call parking, it just plays back voice clips. I can remove /etc/asterisk/modules.conf modules one by one: [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so ;obsolete noload = chan_alsa.so noload = chan_oss.so noload = chan_skinny.so noload = chan_phone.so noload = app_voicemail.so noload = chan_zap.so noload = app_meetme.so But I've not made it very far building up just what I need: [modules] autoload=yes load = res_crypto.so load = res_features.so load = chan_iax2.so load = chan_sip.so load = codec_gsm.so load = codec_ulaw.so I get hard to track down symbol errors like: [chan_iax2.so]Aug 21 10:37:02 WARNING[16384]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_check_signature Aug 21 10:37:02 WARNING[16384]: loader.c:374 load_modules: Loading module chan_iax2.so failed! [chan_iax2.so]Aug 21 10:38:12 WARNING[16384]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_moh_stop Does anyone know a better way to do this? Grep'ing the soruce by trial and error is not working. Modules have to be in just the right order. -Bryce ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 DTMF not recognized - Bug report - Help sought
I have working SIP numbers with broadvoice, and just added a DID from
http://connect.voicepulse.com/ . The calls answer, but DTMF is not
recognized.
With iax2 debug active pressing DTMF does nothing. Zilch. Zero.
A friend tried a different IAX2 connection, and got the same results.
I see the following in the archives:
On Fri, 2004-04-09 at 10:12, Robert Jackson wrote:
Hey all,
I am dialing a DID through VoicePulse Connect. The number is
answered by a main menu type of IVR. The configuration is as specified
in both the wiki and VoicePulses documentation. The call comes through
without a problem, but when the caller enter any keys they are either
not recieved by * or they are ignored. With SIP I would typically put a
dtmfmode= line under the peer and everything works great, but I am not
sure how to attack this. I found a few items referring to the same
issue in the list, but I didn't find any answers. If this is a bug I
will create a report on the bugtracker, but I would rather make sure
that I am not just completely dense and not seeing the easy answer. I'm
trying to replicate the issue with NuFone.
CVS from 2004-04-04 stable branch.
JC wrote on Wed, 28 Jan 2004 19:47:41 -0500
Hello all, I am using voicepulse DID's to receive calls via IAX to and =
asterisk IVR dial plan I have put together. The problem is after 3-5mins =
the system cant pickup the DTMF tones I am sending... I have tried =
different telephones... It just repeats menu options over and over I =
have to call back and then it works again for another few mins...
Any ideas... iax.conf? issue?
Thanks,
J.C.
Chris,
Thank you for contacting VoicePulse.
Our engineers are aware of the DTMF problem and are working to have it
resolved as quickly as possible.
Please reply directly to this email if we can provide any additional
assistance.
Regards,
VoicePulse Customer Support
I'm running:
/usr/src/asterisk/asterisk -r
Asterisk CVS-HEAD-08/01/04-22:51:56, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-HEAD-08/01/04-22:51:56 currently running on skip (pid = 2522)
skip*CLI
My /etc/asterisk/extensions.conf does:
exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
[voicepulse-incoming]
exten=510740,1,Ringing
exten=510740,2,Wait,3
exten=510740,3,Answer
exten=510740,4,Agi,/usr/local/mipl/agnese|http://www..com/X.cgi?source=${EXTEN}callerid=${CALLERIDNUM}
exten=510740,5,Hangup
+++
Is there anyone else with a similar problem? A working setup?
-Bryce
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[Asterisk-Users] Re: Analog FXO Card
-= On 15 Sep 2003 11:09:38 -0600, tom [EMAIL PROTECTED] said: And interestingly, the Digium card looks a lot like a product sold by Tigerjet, called the Personal Phone Gateway. I'm purely speculating on this, but Digium could have used Tigerjet's reference design for their own board. Steve Haehnichen replied: That's kindof how the industry goes. No point in rehashing designs or trying to beat volume manufacturers at their own game. The FCC Reg# on the board is for AMIGO Technology Co of Taiwan. I'm guessing the FXO board is a lot like an AMI-IA92: http://www.amigo.com.tw/products/modem/AMIIA92_IE92.htm You can zoom in here: http://www.amigo.com.tw/catalogue/Modem.pdf The same right down to the AMI-IA92/IE92 on the FXO silkscreen. :) For the record: I bought an XP 100 so that I could too get the support that I expect that I will need. However, I like to know what I buy when I purchase hardware. I do not understand what is this notion of AMI-IA92 - this is being labeled as Intel's software base solution for a V.92 modem under windows. However, this is still showing up in my /proc/pci as a TigerJet 300 Communications Controller. Does this mean that Intel software works for the TigerJet 300? If I boot into windows, could I use this board as a modem? What about T.38 and Fax support for this board, is this envisionable? What I am interested in knowing is whether the sound i/o on this board is down through PCI DMA or its being done through a serial port on a PCI bus. -=Francois=- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Params on SIP URI REGISTER/INVITE
We're doing some SIP development and have a question on additional parameters supplied to the register (in this case maddr= and the non-standard clport= in our example below). What we're experiencing is the INVITE doesn't included these parameters and they get dropped when the INVITE is sent to the 10.1.1.97 address. Ideas? Supported? SIP Bug? REGISTER sip:test1.mydomain.com SIP/2.0 Via: SIP/2.0/UDP10.1.1.97:5060;branch=z9hG4bKd1f1eb5acc28043b83a28ca2ee1e5f15,SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bk-8c166b93 From: JohnDoe sip:[EMAIL PROTECTED];tag=6f0ecbcb3a5e62c4 To: JohnDoe sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Max-Forwards: 69 Contact: JohnDoesip:[EMAIL PROTECTED]:5060;maddr=192.168.0.2;clport=5061;expires=3600 User-Agent: Sipura/SPA2000-1.0.15 Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is asterisks the best for a simple DTMF response system?
I received a recommendation to check out Asterisk, as a platform to host a simple DTMF response system, something like: Setup up VoIP endpoint on Linux/FreeBSD system Answer incoming VoIP phone calls User enters 100#, perl script plays back foo User enters 101#, perl script plays back fum User enters 102#, perl script looks up something in database, converts to text with festival, speaks it. How would one get started, using Asterisks on this project, and is Asterisks the best option? Is it really good enough for a high volume (though sub carrier-grade) solution? I'm willing to use commercial software also. -Bryce ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Plugging Asterisk Security Holes....
Asterisk works fine across cipe tunnels, quite happily got IAX links running to my home from work over a cipe link. You probably won't get ssh port forwarding running because IAX uses udp and I think ssh only forwards tcp by default. Date: Tue, 23 Mar 2004 19:53:46 -0600 (CST) From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Plugging Asterisk Security Holes Reply-To: [EMAIL PROTECTED] Hello, I am interested in knowing if someone has done any work on IPSec VPN SSH port forwarding for Asterisk boxes. If so, it will be nice if we can all share our experiences here. I am perticularly interested in finding out which solution is the best for securing voice channels over the internet. Assuming we use IAX protocol, does it make any difference? Another topic of interest is securing the box itself. Does a firewall (hardware outside of the box or a linux based firewall) suffice the need? Let's discuss some of the security issues around asterisk here. Thanks a lot for your feedbacks and comments. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie seeks help: Getting Asterisk to run on Mandrake 9.2
Hello, i was hoping someone from the list could point me in the correct direction. We recently purchased the Asterisk Developer's Kit (TDM) over at this link: http://www.digium.com/index.php?menu=developerskit_tdm And now, i'm trying to get this to work in Mandrake 9.2. I've gotten a fairly recent source RPM that takes the source from CVS, compilies and links, and provide three neat packages. I've gotten the kernel modules to load, the stock config files are in /etc/asterisk, and asterisk runs when invoked by asterisk -c My problem now is that i can't seem to get a dial tone from the extension phone. (i've connected a phone to the FXS, and an outside line to the FXO). Alhtough the LED next to the phone socket lights up, and the phone earpiece emits a tone when you press a key, there is no dialtone, and no matter what you do, nothing happens = both on the screen/cli of asterisk and the phone itself. Silence. When you call up the direct line, it just rings forever. Now, i'm thinking... config problem. However the stock config files of asterisk are a lot, and i haven't seen a config-set which is tailored to my exact setup. I've tried http://www.voip-info.org/wiki-Asterisk+quickstart But that gets you going with SIP, which i think i'll do when i've gotten the phone extension to dial 9 for an outside line successfully first. Any ideas or pointers to a cookbook recipe would be very much appreciated. Thanks in advance. - Ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] extensions problem
Your phone supports call waiting, so isn't giving out busy. I had the same problem with a budgetone 102, you can't turn this off on the phone but you can work round it by adding Incominglimit=1 Into the sip.conf entry for the phone From: Jon Lawrence [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Mon, 15 Mar 2004 15:29:01 + Subject: [Asterisk-Users] extensions problem Reply-To: [EMAIL PROTECTED] Hi, I've got 2 x100p's installed in my system. Both execute the same incoming contexts as follows: [inboundA] include = dialjon [inboundB] include = dialjon|09:00-16:30|Mon-Fri|*|* [dialjon] exten = s,1,answer exten = s,2,Dial(SIP/2000,15) exten = s,3,Playback(noone) exten = s,103,Goto(onphone,s,1) snip Am I right in saying: if no one answers at ext 2000 then s,3 is executed. if ext 2000 is busy then 103 is executed. If so then sometihng is wrong. If I'm already on a call, I want 103 to be executed however, this isn't happening. If a new call comes in (whilst I'm talking on ext 2000) I here it ringing on my handset. Can anyone point out where I've gone wrong ? TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pingtel SIPxchange IP PBX goes Open Source...
I just read that Pingtel (www.pingtel.com) will be releasing it's IP PBX (which runs under Linux) to open source (similar model to Redhat Linux, charging for support, etc.). Read more about it at... http://www.pingtel.com/a_opensource.jsp and http://www.tmcnet.com/usubmit/2004/Feb/1024036.htm I love Asterisk, I've migrated my entire company over to it ... maybe we can gleam some technology from this new Open Source Project. I have no idea how SIPxchange ranks up with other IP PBX products. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gastman doesn't draw lines properly between resources ...
I really like the functionality that Gastman provides, it would solve a problem I currently have that Secretaries don't know when/who's on the phone before they transfer the caller... But I'm seeing some oddities, maybe just because the code line hasn't been updated in a while. Take a look at the following images: http://www.voiping.com/asterisk/gastman1.jpg http://www.voiping.com/asterisk/gastman2.jpg http://www.voiping.com/asterisk/gastman3.jpg Basically I *thought* that it would draw from the icons to the resources. It seems to dynamically generate new icons (with the same images) and make them randomly appear It does change the led indicator ... but I'd like it not to draw _new_ icons, and draw a line between the proper resources? In some cases, like in gastman1.jpg it drew a new icon *and* a connection to the conference room icon too? gastman3 shows extension 24 connected to voicemail but drew a new icon for it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Current version of gastman precompiled binary
Looking for a current precompiled Win32 binary for gastman, don't have a build environment for Windows. Also does gastman compile under Linux and is there a current binary as well... Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reorder tone ...when it should be Busy...
I've noticed I have an issue with my Dialplan ... apparently instead of a busy
signal when the caller is busy it falls through and gets a Congestion...
What's the proper syntax for this, reorder tone when there is a reorder and
busy when there is a busy...
SBC is a T1/PRI.
[macro-sbc-outdial]
exten = s,1,Dial(${ARG1}/${ARG2})
exten = s,2,Congestion
exten = s,102,Playback(noservice)
exten = s,103,Congestion
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[Asterisk-Users] Remote reloading Cisco phones...
Here's a simple small expect script ... I call it phreboot, usage: phreboot IP $ phreboot 10.99.1.1 -- cut here -- #!/usr/bin/expect -f set timeout -1 spawn $env(SHELL) match_max 1 send -- telnet [lrange $argv 0 0]\r expect -exact word : send -- cisco\r expect -exact Phone send -- reset\r send -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using ACD functionality for main number answer and music on hold
I'm considering using the Agent login/logoff function to add to a queue that will be our main number during the day to answer. Periodically our receptionist is not at her desk and would be useful for her to login elsewhere and get the main number calls to transfer as she sees fit. If the agent's don't pick up in a specific amount of time, it's transferred to our main IVR... I have the functionality working, but right now when you dial the main number you get the musiconhold that is defined for that queue. Is there a way (short of recording a mp3 of a ringing phone) for the person to get a ringing sound instead of the MOH? Thanks, Lenny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need Cisco 7940 or 7960s at good price for Asterisk deployment
Folks -- I know this isn't directly an * issue, but I need to buy 14 7940s (preferably) (or 7960s if the price is also reasonable) --- no power cubes, immediately. If anyone has a good price, contact me offline at 512-427-1324 or lenny @ rocksteady.com Thanks, Lenny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Play a sound after dialing a user...
I'd like to play a sound to a user I dial (via SIP) once they answer play the sound and then allow me to talk to them. The new Cisco 7960 SIP code allows to set lines to autoanswer via the speaker phone, I'd like to play a tone after it rings through and then talk... Any thoughts on how to do this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Latest CVS causes compile time error
gcc -shared -Xlinker -x -o chan_phone.so chan_phone.o gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DASTERISK_VERSION=\CVS-03/25/03-10:49:30\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -Wno-missing-prototypes -Wno-missing-declarations -DIAX_TRUNKING -DCRYPTO -o chan_zap.o chan_zap.c chan_zap.c: In function `zt_digit': chan_zap.c:677: warning: implicit declaration of function `pri_information' chan_zap.c:677: structure has no member named `pri' chan_zap.c:677: structure has no member named `call' chan_zap.c: In function `zt_call': chan_zap.c:1144: warning: unused variable `s' make[1]: *** [chan_zap.o] Error 1 Yes, I have the latest zaptel too. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH w/SIP (Cisco 7960) error received.
-- Registered SIP '' at 192.70.239.2 port 5060 expires 3600 -- Executing Playback(SIP/lenny-4ee2, transfer|skip) in new stack -- Executing Macro(SIP/lenny-4ee2, dial|7555|SIP/lenny-lap) in new stack -- Executing Dial(SIP/lenny-4ee2, SIP/lenny-lap|20|tT) in new stack -- Called lenny-lap -- SIP/lenny-lap-92fb is ringing -- SIP/lenny-lap-92fb answered SIP/lenny-4ee2 -- Attempting native bridge of SIP/lenny-4ee2 and SIP/lenny-lap-92fb -- Started music on hold, class 'default', on SIP/lenny-lap-92fb WARNING[20501]: File chan_sip.c, Line 796 (sip_write): Asked to transmit frame type 64, while native formats is 4 (read/write = 8/4) -- Stopped music on hold on SIP/lenny-lap-92fb == Spawn extension (macro-dial, s, 1) exited non-zero on 'SIP/lenny-4ee2' in macro 'dial' == Spawn extension (default, s, 2) exited non-zero on 'SIP/lenny-4ee2' What's that error above? I can play the MOH MP3 file with the MP3Player app just fine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lastest CVS built compile time error
ZT_SIG_SF undeclared? make[1]: Entering directory `/usr/local/src/asterisk/channels' gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DASTERISK_VERSION=\CVS-03/12/03-21:24:47\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -Wno-missing-prototypes -Wno-missing-declarations -DIAX_TRUNKING -DCRYPTO -o chan_zap.o chan_zap.c chan_zap.c: In function `sig2str': chan_zap.c:784: `ZT_SIG_SF' undeclared (first use in this function) chan_zap.c:784: (Each undeclared identifier is reported only once chan_zap.c:784: for each function it appears in.) chan_zap.c: In function `zt_call': chan_zap.c:1235: `ZT_SIG_SF' undeclared (first use in this function) chan_zap.c:1131: warning: unused variable `s' chan_zap.c: In function `zt_answer': chan_zap.c:1675: `ZT_SIG_SF' undeclared (first use in this function) chan_zap.c: In function `zt_handle_event': chan_zap.c:2434: `ZT_SIG_SF' undeclared (first use in this function) chan_zap.c: In function `zt_new': chan_zap.c:3363: `ZT_SIG_SF' undeclared (first use in this function) chan_zap.c: In function `ss_thread': chan_zap.c:3527: `ZT_SIG_SF' undeclared (first use in this function) chan_zap.c: In function `handle_init_event': chan_zap.c:4164: `ZT_SIG_SF' undeclared (first use in this function) chan_zap.c: In function `mkintf': chan_zap.c:4711: `ZT_SIG_SF' undeclared (first use in this function) chan_zap.c: In function `load_module': chan_zap.c:6273: `ZT_SIG_SF' undeclared (first use in this function) make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/local/src/asterisk/channels' make: *** [subdirs] Error 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
