Re: [asterisk-users] Best Firewall Suggestions?
Hello, we are using vyatta, a linux based router. the software is more focused on routing capabilities, than on firewall rules, but it works fine an there is a very good support. for ha you can use it in a cluster. bye, patrick -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Tel. +49 2151 5554-263 Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI Dummy for Linux VServers
I'm running dahdi on the host system, and have added the /dev/dahdi/ devices to the guest vserver as recommended in Beave's Virtual Private Asterisk whitepaper (http://www.telephreak.org/papers/vpa/). I tried copying libtonezone.so and libtonezone.h to the guest, but I couldn't anything to replace zaptel.h in DAHDI souce (it seems dahdi.h was deprecated?). I need to fool asterisk configure script into thinking dahdi was actually installed because ./configure script isn't satisfied with just having the devices present. Any suggestions for getting asterisk to build with support for components using DAHDI timers in vservers (for meetme, internal timing, etc.)? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in the Cloud
On Oct 13, 2009, at 5:22 PM, Dan Journo wrote: Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I’d appreciate it. Many thanks Dan Journo Dan, I'm giving a presentation at AstriCon on this very topic. Here's a writeup of our experience with Asterisk on Amazon EC2 http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405 . -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXS to SIP gateway
Hello list ! I don't have the money to test out all the products and reading the manuals is not always that enlightening... Does someone here know a good gateway-product that lets analogue telephones communicate with an Asterisk-server. I have found the Grandstream GXW-400x to be able to add SIP-accounts to analogue telephone devices that are connected to the FXS-ports. Moreover this product has a backup-PSTN line for emergency calls and backup. Could you advice other products/manufacturers ? Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in the Cloud
Thanks Eric, I'd love to be able to make it to an Astricon one day. At the moment, its a bit out of my price range. Do you happen to know whether RackspaceCloud.com offers a Kernel with a timing device enabled? Many thanks and good luck with the presentation. Dan -Original Message- From: Eric Chamberlain e...@rf.com Sent: 14 October 2009 08:10 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk in the Cloud On Oct 13, 2009, at 5:22 PM, Dan Journo wrote: Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, Id appreciate it. Many thanks Dan Journo Dan, I'm giving a presentation at AstriCon on this very topic. Here's a writeup of our experience with Asterisk on Amazon EC2 http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405 . -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple call
Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager Thanks and regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in the Cloud
I've managed to install and run a successful test using RackspaceCloud. I didn't know how else to test the timing device so I just placed a 3 way call which worked fine. My question is, how can I actually test the timing device to see if it is suitable? Many thanks Dan Journo -Original Message- From: Eric Chamberlain e...@rf.com Sent: 14 October 2009 08:10 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk in the Cloud On Oct 13, 2009, at 5:22 PM, Dan Journo wrote: Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, Id appreciate it. Many thanks Dan Journo Dan, I'm giving a presentation at AstriCon on this very topic. Here's a writeup of our experience with Asterisk on Amazon EC2 http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405 . -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP - busy tone detection
Hi guys, My Asterisk box is connected to a VoIP provider using SIP protocol. Unforunately, when I try to call a number that is busy, the provider plays busy tones for 15 seconds and than hungups the call with dialstatus=CHANUNAVAIL, instead of dialstatus=BUSY. Is there any way to detect busy tone pattern on a SIP call (are there any SIP equvalents of busydetect and busycount parameters)? Thanks, Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF failing in some calls
Hello, I am using Asterisk 1.2.33 under Debian ETCH linux. I have the following problem with DTMF: In my callback system, I calls an access DID. My system calls me back to my phone. It asks me for a password to let me dial an international number. If the authentication succeeds, I can dial a number in the system. Sometimes Asterisk catches only some of the digits I have entered. Sometimes, it duplicates some other digits. Sometimes, the system does not catch anything. And finally, sometimes it works properly. I am using rfc2833 as dtmf mode. Please help. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS to SIP gateway
jonas kellens wrote: Hello list ! I don't have the money to test out all the products and reading the manuals is not always that enlightening... Does someone here know a good gateway-product that lets analogue telephones communicate with an Asterisk-server. I have found the Grandstream GXW-400x to be able to add SIP-accounts to analogue telephone devices that are connected to the FXS-ports. Moreover this product has a backup-PSTN line for emergency calls and backup. Could you advice other products/manufacturers ? I hope to see more replies because I was in your situation some years ago. I'm far from an expert, but in my experience, at _that_ price range you don't have a lot of products to choose from, the Cisco SPA3102 is similar to what you are describing (Plus it's also a PSTN GW). Of this kind, I used GXW-4004 and Linksys PAP2, SPA3000 (the Cisco 3102 predecesor), also some larger AddPac ATAs (www.addpac.com) with excellent results, all of them have their pros and cons. The Digium TDM410 cards and they have a very good price/quality relation, plus they are intended for asterisk, plus you are supporting asterisk development. For an ATA (FXS only) list you can check http://www.voip-info.org/wiki/view/Analog+Telephone+Adapters -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing sip calls work; incoming calls fail
listm...@websage.ca wrote: On Sat, 10 Oct 2009 18:02:04 -0700 listm...@websage.ca wrote: On Sun, 11 Oct 2009 02:11:47 +0200 Ivan Stepaniuk i...@albafotonica.com wrote: listm...@websage.ca wrote: On the LAN side I can see the INVITE and OKAY messages which end with a CANCEL, apparently initiated by the Asterisk gateway. On the WAN side I can see that my Asterisk gateway is repeatedly sending OKAY messages in response to the INVITE from my ITSP. I assume the trouble is that these messages are either not getting back to my provider or something is blocking the confirmation from them. This more or less confirms what was seen in the sip debug trace as well. Post that SIP message from the CLI (sip debug), try adding externip=XXX.XXX.XXX.XXX (your external/public IP address) to your sip.conf global section, asterisk may be including it's private address in the OKAY sent to your provider. Here's the last message in sip debug before it gives up: ... Retransmitting #6 (no NAT) to 66.51.127.173:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bKee03.6a63373.0;received=66.51.127.173 Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKpae7KcSeB2Bte Record-Route: sip:66.51.127.173;lr;ftag=9Z5N4eayXp3Qm From: 2508864577 sip:2508864...@66.51.127.163;tag=9Z5N4eayXp3Qm To: sip:12504129...@66.51.127.173;tag=as32af6364 Call-ID: b734bd26-2fae-122d-91a5-653b331e358a CSeq: 121440337 INVITE User-Agent: Asterisk PBX 1.6.0.15 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: sip:12504129...@96.50.76.138 Content-Type: application/sdp Content-Length: 262 v=0 o=root 992672626 992672626 IN IP4 96.50.76.138 s=Asterisk PBX 1.6.0.15 c=IN IP4 96.50.76.138 t=0 0 m=audio 15550 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical packet (see doc/sip-retransmit.txt). Scheduling destruction of SIP dialog '4e8ef1b977bf0e062212334634080...@192.168.11.1' in 6400 ms (Method: INVITE) ... 66.51.127.173 is my provider's SIP server 66.51.127.163 is my provider's RTP server I even check DNS to make sure both forward and reverse records jive. Externip was a good suggestion, and worth a try, though because I'm registering with my provider and using dynamic=yes, wouldn't they just reply to that anyway, especially given that the registration works fine? Anyway, after adding externip=my-external-ip to [general] and doing a sip reload in the console the problem remains... [Bumping this in the hope that someone might have some new insight or suggestions since I posted this on a holiday weekend (in my part of the world anyway)...] I was staring at the SIP transcript and I don't see anything wrong, I'm out of suggestions, except that you could analyze and compare the packets when your phone is connected directly (if it's physically possible). I hope someone throws some light over this. -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP - busy tone detection
Asterisk wrote: Hi guys, My Asterisk box is connected to a VoIP provider using SIP protocol. Unforunately, when I try to call a number that is busy, the provider plays busy tones for 15 seconds and than hungups the call with dialstatus=CHANUNAVAIL, instead of dialstatus=BUSY. Is there any way to detect busy tone pattern on a SIP call (are there any SIP equvalents of busydetect and busycount parameters)? Thanks, Alex AFAIK this is not possible, yet. See this post, about the same topic: http://archives.free.net.ph/message/20090703.122514.65031d43.en.html -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
Gordon Henderson gordon+aster...@drogon.net writes: I use Draytek Vigor 2820's these days. Mostly (when not having something more corporate or dealing with geeks who want a Linux based one) Built in hardware assist VPN too. They do have a SIP ALG, but it's turned off by default (the earlier ones had it turned on) Port forwarding works as you'd expect it to, and the traffic shaping is better than no traffic shaping. Well hopefully Draytek's are now better, but when I tried to use them 2 years ago the software was complete crap. VPN was unreliable (and supported only one tunnel per gateway), DHCP relay broken, DHCP server broken and lacking features, dynamic DNS implementation got them blacklisted by OpenDNS... /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP RealTime defaultuser Field Cleared
Hi All, I am running Asterisk 1.6.1.2 with realtime for SIP and recently noticed the values in the defaultuser field have been disappearing. I can't place my finger on what is happening but it appears that when the peer de-registers the defaultuser field is cleared. I will be having a more detailed look through the logs and possibly add more logging to the database but wondered if anyone else has had this problem. Kind Regards Stuart Elvish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy on asterisk 1.6
hey In 1.6 version actually not wrote any code for option 'o'
you need to add following line into file
Index: apps/app_chanspy.c
===
--- apps/app_chanspy.c (revision 215998)
+++ apps/app_chanspy.c (working copy)
@@ -427,7 +427,12 @@
return -1;
}
- f = ast_audiohook_read_frame(csth-spy_audiohook, samples,
AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR);
+ if (ast_test_flag(chan, OPTION_READONLY)) {
+ /* Option 'o' was set, so don't mix channel audio */
+ f = ast_audiohook_read_frame(csth-spy_audiohook, samples,
AST_AUDIOHOOK_DIRECTION_READ, AST_FORMAT_SLINEAR);
+ } else {
+ f = ast_audiohook_read_frame(csth-spy_audiohook, samples,
AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR);
+ }
ast_audiohook_unlock(csth-spy_audiohook);
regards
Dhaval
2009/10/14 Jorge Gutiérrez jgutier...@palosanto.com
I have read about that on asterisk 1.6, there will be a parameter o (Only
listen to audio coming from this channel), I have tried, but I still get
inbound and outbound audio from the spied channel.
Has anyone used this feature? Is it working? Is there any work-around?
I will like to only spy the outbound audio from a channel, I dont want to
hear the incomming audio of that channel.
I have used the following context:
[Conf]
exten = s,1,Answer
exten = s,2,Background(custom/menu_test)
exten = s,3,ChanSpy(,qoX)
exten = 1,1,Goto(Conf,s,2)
exten = 2,1,AGI(conf.php,${CALLERID(num)},${SPY_CHANNEL})
exten = 2,n,Goto(s,3)
exten = s,n,Goto(test2,s,1)
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[asterisk-users] Queues with unavailable members
We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone=SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a member of a queue. In that case Queue() keeps sending the call to the phone, and the cell-phone=SIP gateway dutifully makes a call, which is then rejected by the cellular network. A few seconds later, Queue() tries again. This needlessly wastes resources both in Asterisk and in the cellular network. One idea is to run the call through chan_local (we do this anyway because we need to format the caller-ID differently for different phones) and then record if a call is rejected, and for the next 30 seconds just abort if we are asked to send a call to that particular phone. The downside is that we are still running a call through part of the dial plan, but at least it can be done in perhaps 3 lines of code. I would very much like to hear about smarter ways to do it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 vs 1.6
Hi, I was wondering whether there are any problems with v1.6 which means I should avoid it. What are the advantages of upgrading? And finally, why both versions are available? Why not just scrap 1.4? Many thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
You could configure them as agents and have them log off automatically after a while they're not responding. l. 2009/10/14 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone=SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a member of a queue. In that case Queue() keeps sending the call to the phone, and the cell-phone=SIP gateway dutifully makes a call, which is then rejected by the cellular network. A few seconds later, Queue() tries again. This needlessly wastes resources both in Asterisk and in the cellular network. One idea is to run the call through chan_local (we do this anyway because we need to format the caller-ID differently for different phones) and then record if a call is rejected, and for the next 30 seconds just abort if we are asked to send a call to that particular phone. The downside is that we are still running a call through part of the dial plan, but at least it can be done in perhaps 3 lines of code. I would very much like to hear about smarter ways to do it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 vs 1.6
On Wed, Oct 14, 2009 at 9:01 AM, Dan Journo d...@keshercommunications.com wrote: I was wondering whether there are any problems with v1.6 which means I should avoid it. Try searching the list for the many times this has been answered. Since this is your choice, you need to set up a parallel instance of your environment and vet your particular usage. At the very least you will need to update your dialplan to the new syntax, and upgrade to DAHDI if you're using hardware phone cards. What are the advantages of upgrading? Features, both to the individual applications and new applications not previously available. Scalability, especially for large dialplans, and a much better SIP stack. More eyes on the code as it's the current track. Try searching the list for the many times this has been answered. And finally, why both versions are available? Why not just scrap 1.4? 1.2 is still available too. Because the choice is yours. Many people are still using 1.2, much less 1.4. Some people call old code 'stable' code, as in the bugs are known or worked around. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no outbound calls
here is the debug from the CLI. I think I know where the problem is I just can figure out how to fix it. The IP in the From and To i think is where the problem is. When I make an outbound call. i get the message the call cannot be completed as dialed. if i call another ext it works. I posted the debug for both calls. ==outbound call=== --- Transmitting (NAT) to 10.0.0.46:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKfd2143ede5319cf9b.273e4b904b0cc3101;received=10.0.0.46 From: ext sip:1...@10.0.0.8;tag=9d9e3944ba To: 93214545 sip:93214...@10.0.0.8;tag=as290bd498 Call-ID: 401d30b0a1893e80 CSeq: 13401 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:99676...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 14398 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv = ext to ext=== SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK280378de3608b5bf1.2576ced30c6198b74;received=10.0.0.46 From: ext sip:1...@10.0.0.8;tag=d729237fcc To: 111 sip:1...@10.0.0.8;tag=as553ab5e9 Call-ID: c7cc32657c620790 CSeq: 8007 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:1...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 10414 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
What is the command to log off the agents ? Thx On Wed, Oct 14, 2009 at 6:45 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: You could configure them as agents and have them log off automatically after a while they're not responding. l. 2009/10/14 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone=SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a member of a queue. In that case Queue() keeps sending the call to the phone, and the cell-phone=SIP gateway dutifully makes a call, which is then rejected by the cellular network. A few seconds later, Queue() tries again. This needlessly wastes resources both in Asterisk and in the cellular network. One idea is to run the call through chan_local (we do this anyway because we need to format the caller-ID differently for different phones) and then record if a call is rejected, and for the next 30 seconds just abort if we are asked to send a call to that particular phone. The downside is that we are still running a call through part of the dial plan, but at least it can be done in perhaps 3 lines of code. I would very much like to hear about smarter ways to do it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple call
kaustuva...@bbsr.syscomes.com escribió: Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager Thanks and regards Hi, Use the AMI Originate Action with the Async option and set an unique ActionID to track the response for each one: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate If you're writing an external application, using some open source libraries will make your work easier: http://phpagi.sourceforge.net/ http://asterisk-java.org/ Or look for a good one made on your favorite programming language. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Have you tried autopause=yes in your queue configuration? You can then unpause the member by either the dialplan (e.g. having the cell phone user log back in) or using an AMI based program to change the paused state. You can read more about the latter here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html -Elliot -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen Sent: Wednesday, October 14, 2009 7:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues with unavailable members We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone=SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a member of a queue. In that case Queue() keeps sending the call to the phone, and the cell-phone=SIP gateway dutifully makes a call, which is then rejected by the cellular network. A few seconds later, Queue() tries again. This needlessly wastes resources both in Asterisk and in the cellular network. One idea is to run the call through chan_local (we do this anyway because we need to format the caller-ID differently for different phones) and then record if a call is rejected, and for the next 30 seconds just abort if we are asked to send a call to that particular phone. The downside is that we are still running a call through part of the dial plan, but at least it can be done in perhaps 3 lines of code. I would very much like to hear about smarter ways to do it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is intended only for the use of the individual (s) or entity to which it is addressed and may contain information that is privileged, confidential, and/or proprietary to Calling Circles LLC and its affiliates. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, forwarding or copying of this communication is prohibited without the express permission of the sender. If you have received this communication in error, please notify the sender immediately and delete the original message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Dummy for Linux VServers
Try installing DAHDI from source in the guest, and instead of starting it as usual try fooling Asterisk with the /dev hack you did. That way you would have all the dependencies for compiling Asterisk and could still use the devices you made available in /dev. Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP - kn0x atlanticny...@gmail.com escreveu: I'm running dahdi on the host system, and have added the /dev/dahdi/ devices to the guest vserver as recommended in Beave's Virtual Private Asterisk whitepaper (http://www.telephreak.org/papers/vpa/). I tried copying libtonezone.so and libtonezone.h to the guest, but I couldn't anything to replace zaptel.h in DAHDI souce (it seems dahdi.h was deprecated?). I need to fool asterisk configure script into thinking dahdi was actually installed because ./configure script isn't satisfied with just having the devices present. Any suggestions for getting asterisk to build with support for components using DAHDI timers in vservers (for meetme, internal timing, etc.)? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Config Files
Greetings, I have a fresh asterisk installation. When I install I get all of the config files. What is the best way to get a 'stripped' down system with just the bare config files I would need to do a sip connection? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy
I am unsing Asterisk 1.2.28 I want please to use ChanSpy urgently my /etc/asterisk/extensions_additional.conf is as follow: [chanspy] include = chanspy-custom exten = 102**,1,Chanspy(102) exten = 102**,n,Hangup exten = 103**,1,Chanspy(103) exten = 103**,n,Hangup exten = 400**,1,Chanspy(400) exten = 400**,n,Hangup exten = 501**,1,Chanspy(501) exten = 501**,n,Hangup exten = 601**,1,Chanspy(601) exten = 601**,n,Hangup exten = 606**,1,Chanspy(606) exten = 606**,n,Hangup ; end of [chanspy] I created a Context to put my extension into it to be able to use ChanSpy. While there is a call with an extension 102 and my extension is 606 i call 102** to spy but i couldn't hear anything, all i hear is beep -- Executing ChanSpy(SIP/606-09430fd0, 102) in new stack -- Playing 'beep' (language 'en') -- Playing 'beep' (language 'en') Thanks Torintino _ Windows Live: Make it easier for your friends to see what you’re up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Dummy for Linux VServers
On Wed, Oct 14, 2009 at 10:56:05AM -0300, Vinícius Fontes wrote: Try installing DAHDI from source in the guest, You should also probably bind-mount /dev/dahdi from the hosts to the guests . Or just create those device files for the guests somehow. Alternatively, try using Asterisk = 1.6.1 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS to SIP gateway
On Wed, Oct 14, 2009 at 12:27 AM, jonas kellens jonas.kell...@telenet.be wrote: Hello list ! I don't have the money to test out all the products and reading the manuals is not always that enlightening... Does someone here know a good gateway-product that lets analogue telephones communicate with an Asterisk-server. I have found the Grandstream GXW-400x to be able to add SIP-accounts to analogue telephone devices that are connected to the FXS-ports. Moreover this product has a backup-PSTN line for emergency calls and backup. Could you advice other products/manufacturers ? We have used Cisco 2800 series routers with voice cards that work fine for PRI, but don't implement SIP the way they should. Analog was not so great. We also tried to convert some Cisco VG-224s to SIP with limited success. I don't recommend using either of those (plus they are expensive...) Grandstream (GXW-4024) had major issues with Fax, so we only use them for connecting for voice only applications. They seem to work well with Asterisk, and are easy to configure. Don't count on fax working at all though, or even worse working in some cases... AudioCodes is where we finally found a product that does what we need. They are about twice as much as a Grandstream (at least for the MP-124 vs GXW-4024) but have been rock solid for faxing so far. They also come in multiple configurations which is handy. We use the MP-114 2FXS/2FXO device at our remote sites for local PSTN access and to connect a fax machine. They also support survivability (proxy registration) in case of WAN failure. The complaints that I have are that the web interface has A LOT going on, and there is no real CLI to speak of. Neither of these are real issues, just takes you a few more minutes up front to read the manual. I haven't tried any Adtran devices but have thought about purchasing one to test with if I ever get the time. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy
You must use extenspy if you want to spy on specific extension. Otherwise you can only cycle through available channels. Regards Rennes -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Torintino T Sent: Wed 10/14/2009 17:46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSpy I am unsing Asterisk 1.2.28 I want please to use ChanSpy urgently my /etc/asterisk/extensions_additional.conf is as follow: [chanspy] include = chanspy-custom exten = 102**,1,Chanspy(102) exten = 102**,n,Hangup exten = 103**,1,Chanspy(103) exten = 103**,n,Hangup exten = 400**,1,Chanspy(400) exten = 400**,n,Hangup exten = 501**,1,Chanspy(501) exten = 501**,n,Hangup exten = 601**,1,Chanspy(601) exten = 601**,n,Hangup exten = 606**,1,Chanspy(606) exten = 606**,n,Hangup ; end of [chanspy] I created a Context to put my extension into it to be able to use ChanSpy. While there is a call with an extension 102 and my extension is 606 i call 102** to spy but i couldn't hear anything, all i hear is beep -- Executing ChanSpy(SIP/606-09430fd0, 102) in new stack -- Playing 'beep' (language 'en') -- Playing 'beep' (language 'en') Thanks Torintino _ Windows Live: Make it easier for your friends to see what you're up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.421 / Virus Database: 270.13.112/2391 - Release Date: 10/13/09 19:11:00 winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple call
You might be able to do multiple calls using originate to a context instead of an application. Can't really try b/c my test box only has one line. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faheem Sent: Wednesday, October 14, 2009 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] multiple call Through Asterisk AMI, you can not dial multiple number at the same time. If you are going to implement a concurrent call scenario, then AMI would not be a valid choice. Multiple calls can be implemented with callfile. Faheem --- On Wed, 10/14/09, kaustuva...@bbsr.syscomes.com kaustuva...@bbsr.syscomes.com wrote: From: kaustuva...@bbsr.syscomes.com kaustuva...@bbsr.syscomes.com Subject: [asterisk-users] multiple call To: asterisk-users@lists.digium.com Date: Wednesday, October 14, 2009, 11:44 PM Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager Thanks and regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Config Files
Here is sample configuration Just add a registry string and a sip account and send all calls to this account. For example in dial string in extensions.conf exten=,_X.,1,Dial(SIP/14168404...@adf) /// ; sip.conf ; Registry string register= adf:1...@voip-provider.com:9060 ;sip account [adf] type=peer host=voip-provider.com port=9060 context=default country=us dtmfmode=rfc2833 restrictcid=no canreinvite=yes insecure=no disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm promiscredir=yes t38_udptl=yes qualify=25000 nat=yes Muhammad Faheem --- On Wed, 10/14/09, Matt mhop...@gmail.com wrote: From: Matt mhop...@gmail.com Subject: [asterisk-users] Config Files To: asterisk-users@lists.digium.com Date: Wednesday, October 14, 2009, 7:39 PM Greetings, I have a fresh asterisk installation. When I install I get all of the config files. What is the best way to get a 'stripped' down system with just the bare config files I would need to do a sip connection? -Inline Attachment Follows- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple call
Through Asterisk AMI, you can not dial multiple number at the same time. If you are going to implement a concurrent call scenario, then AMI would not be a valid choice. Multiple calls can be implemented with callfile. Faheem --- On Wed, 10/14/09, kaustuva...@bbsr.syscomes.com kaustuva...@bbsr.syscomes.com wrote: From: kaustuva...@bbsr.syscomes.com kaustuva...@bbsr.syscomes.com Subject: [asterisk-users] multiple call To: asterisk-users@lists.digium.com Date: Wednesday, October 14, 2009, 11:44 PM Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager Thanks and regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ACD ASR
Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source destination? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ACD ASR
2009/10/14 B.Masoud @ SH i...@saudihome.com Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source destination? Thanks. You can calculate by yourself with cdr's, its only statistics. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PostgreSQL problems
Folks, I know this must be a configuration problem. Just changed servers last nite -- an interim server running 1.6.1.6. Copied all of /etc/asterisk to the new server and fired it up. Now I'm getting: [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:309 pgsql_log: Failed to insert call detail record into database! [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:310 pgsql_log: Reason: ERROR: syntax error at or near ) LINE 1: INSERT INTO cdr ) VALUES ) ^ [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:311 pgsql_log: Connection may have been lost... attempting to reconnect. [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:314 pgsql_log: Connection reestablished. [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:320 pgsql_log: HARD ERROR! Attempted reconnection failed. DROPPING CALL RECORD! [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:321 pgsql_log: Reason: ERROR: syntax error at or near ) LINE 1: INSERT INTO cdr ) VALUES ) any clues appreciated. Thanx, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto Visit my blog at: http://www.pananix.com/cgi-bin/blosxom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PostgreSQL problems
David A. Bandel wrote: Folks, I know this must be a configuration problem. Just changed servers last nite -- an interim server running 1.6.1.6. Copied all of /etc/asterisk to the new server and fired it up. Local or remote PostgreSQL server? Is the new Asterisk server allowed to connect to the PostgreSQL database. Iptables or pg_hba.conf on a remote system. \\||/ Rod -- Now I'm getting: [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:309 pgsql_log: Failed to insert call detail record into database! [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:310 pgsql_log: Reason: ERROR: syntax error at or near ) LINE 1: INSERT INTO cdr ) VALUES ) ^ [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:311 pgsql_log: Connection may have been lost... attempting to reconnect. [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:314 pgsql_log: Connection reestablished. [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:320 pgsql_log: HARD ERROR! Attempted reconnection failed. DROPPING CALL RECORD! [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:321 pgsql_log: Reason: ERROR: syntax error at or near ) LINE 1: INSERT INTO cdr ) VALUES ) any clues appreciated. Thanx, David A. Bandel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco router
I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
Hi Myles, Thanks to you and everyone else that has responded. I've really learned a lot. pFSense and IPCop sounds let best so far for LINUX based firewalls. I'm also wondering if anyone has any suggestions for a standalone firewall appliance like my Linksys WRT54G except one better suited for a small business and that NAT works well with VOIP. Thanks again! David Wathen -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Myles Wakeham Sent: Tuesday, October 13, 2009 9:06 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Best Firewall Suggestions? My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. We're running IPCop (Linux based, open source, 100% free), and its been fantastic for us. www.ipcop.org I spent weeks trialing many others. Even had Astaro send me out a trial box to use. I think we short-listed this down to pfSense, SmoothWall, Astaro and IPCop. Its been a while since we did this, so newer versions might have different test results now, but (if I remember correctly): 1. pfSense - Solid, but was a bit picky on network adapters (we wanted to use a Quad NIC for this). Also was a bit cryptic for setup, but that's probably just us being too lazy to RTFM. 2. Shorewall - this worked out of the box, looked easy to setup, etc. But when it came down to supporting multiple external WAN IP addresses that we had, it fell short and was dismissed as an option. I believe that their commercial version did support this, but had a hard time trying to find who to buy the damn thing from. 3. Astaro - great company to work with. Really helpful, great tech support, etc. Loving all of that. Not loving the $2K+ price tag for what we needed. But then we are stingy and cheap. That's just us. If you have commercial clients, and budget this looked really good. 4. IPCop - its free. Was a dream to install and setup. Support via their mailing list was awesome. The people there didn't make us feel like newbs when we had basic questions to ask. Feature set rivaled all other products, and there is a pretty healthy add-on market for it. QoS was decent, although there are add-ons for better QoS granularity. We chose IPCop. Been running it with Asterisk and our other network apps, servers, etc. for about 4 months straight. Never needed a reboot. Never crashed. Low footprint, and runs on some old dog hardware we had lying around. Like I said, this review is about 6 months old, so things change. That's our biz. Go figure. Of course, your mileage may vary. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
On 10/14/2009 01:29 PM, David Wathen wrote: Hi Myles, Thanks to you and everyone else that has responded. I've really learned a lot. pFSense and IPCop sounds let best so far for LINUX based firewalls. I'm also wondering if anyone has any suggestions for a standalone firewall appliance like my Linksys WRT54G except one better suited for a small business and that NAT works well with VOIP. I use the Secure Computing SG560 (which which recommended by my VOIP provider), and it works very well with IAX2. I haven't tried SIP. Avoid SonicWall. I had bad experiences with their products and VOIP. - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
Depends on what the router is. If you get a 2800 series router (we use 2801s and 2811s for T1s in production with no major issues). You need the T1/E1 module, DSPs, and an IOS that supports voice. For a 2800 series you would need something like: - VWIC2-MFT-T1/E1 ( or VWIC2-2MFT-T1/E1 if you want two ports) - PVDM2-32 (PVDM2-64 if you want two E1 ports, or two PVDM2-32s) - IOS that supports voice (I use spservicesk9) If you are looking at an older router like a 2651XM or something, you will need something like: - NM-HDV2-2T1/E1 (not the VWIC2, as the NM has build in ports) - PVDM2-32 If you have a specific router in mind, I can be more specific. -Jonathan On Wed, Oct 14, 2009 at 11:00 AM, Julian Lyndon-Smith aster...@dotr.com wrote: I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Config Files
At 10:39 AM on 14 Oct 2009, Matt wrote: Greetings, I have a fresh asterisk installation. When I install I get all of the config files. What is the best way to get a 'stripped' down system with just the bare config files I would need to do a sip connection? In my experience, this was a bit of a procedure. But I thought it was worth it to reduce clutter. Basically, you need to find out which modules you need, and set autoload=no in modules.conf, and then load (with load=xxx.so) each module individually, making sure you've also loaded each module's dependencies. Finding these dependencies was a bit of leg work. I don't remember where I figured that out... sorry. Then you need to find out which modules need which configuration files, and then delete the rest of them... To get you started, I've attached my modules.conf (with sparse comments), and here's a list of the files in my /etc/asterisk directory: asterisk.conf cdr.conf cdr_custom.conf extensions.ael extensions.conf features.conf indications.conf logger.conf modules.conf musiconhold.conf queues.conf sip.conf voicemail.conf zapata-channels.conf zapata.conf YMMV, HTH, HAND. :-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 modules.conf Description: Binary data signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy
Thanks for your reply. Is ExtenSpy available in Asterisk 1.2? If yes, please how can i use it? and how can i cycle through the available channels by ChanSpy? Thanks. Torintino Date: Wed, 14 Oct 2009 18:15:29 +0300 From: rennes.n...@norby.ee To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] ChanSpy You must use extenspy if you want to spy on specific extension. Otherwise you can only cycle through available channels. Regards Rennes -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Torintino T Sent: Wed 10/14/2009 17:46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSpy I am unsing Asterisk 1.2.28 I want please to use ChanSpy urgently my /etc/asterisk/extensions_additional.conf is as follow: [chanspy] include = chanspy-custom exten = 102**,1,Chanspy(102) exten = 102**,n,Hangup exten = 103**,1,Chanspy(103) exten = 103**,n,Hangup exten = 400**,1,Chanspy(400) exten = 400**,n,Hangup exten = 501**,1,Chanspy(501) exten = 501**,n,Hangup exten = 601**,1,Chanspy(601) exten = 601**,n,Hangup exten = 606**,1,Chanspy(606) exten = 606**,n,Hangup ; end of [chanspy] I created a Context to put my extension into it to be able to use ChanSpy. While there is a call with an extension 102 and my extension is 606 i call 102** to spy but i couldn't hear anything, all i hear is beep -- Executing ChanSpy(SIP/606-09430fd0, 102) in new stack -- Playing 'beep' (language 'en') -- Playing 'beep' (language 'en') Thanks Torintino _ Windows Live: Make it easier for your friends to see what you're up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.421 / Virus Database: 270.13.112/2391 - Release Date: 10/13/09 19:11:00 _ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WaitForSilence doesn't work unless Background is called?
I'be been having problems with the WaitForSilence application in my dialplan. The behavior I've noticed, and hope to demonstrated with the fallowing examples, is that WaitForSilence (WFS for short) does not work until Background is called. Note that the lenghts of the sounds are approximately 5 seconds for bring, and 10 for presses. [Testing] exten = s,1,Answer() exten = s,n,Wait(10) exten = s,n,WaitForSilence(5000) exten = s,n,Background(arstandard/bring) exten = s,n,WaitForSilence(5000) exten = s,n,Background(arstandard/presses) I am making constant noise during this dialplan. bring plays, despite the constant noise, indicating that the first WaitForSilence call doesn't work, but the second does work because presses never plays. The Asterisk CLI output can be found here http://pastebin.com/m202fc48e , for this particular dial plan. Other dialplans demonstrate similar behavior. [Testing] ;exten = s,1,Answer() ;exten = s,n,Wait(10) ;exten = s,n,WaitForSilence(5000) exten = s,1,Background(arstandard/bring) exten = s,n,WaitForSilence(5000) exten = s,n,Background(arstandard/presses) I am making constant noise during this dialplan. bring plays instantly, and I can clearly hear the beginning of the message; the dialplan begins at the correct time, and answer detection is working properly. presses never plays, due to the constant noise. WFS works after a call to background but not before. I believe this is related to the lines we are using. By lines I mean the fallowing: g711 codec, SIP protocol, over DSL lines, with Qwest IPLD as the provider. Regular voice conversations are possible over the Qwest lines, and DTMF detection works well. We have tested multiple phones including a couple of our cell phones with various carriers. Any phone we call using the Qwest lines demonstrates the same behavior. When I call locally, thus taking the Qwest lines out of the picture, WFS works as expected. By call locally, I mean that I call from Asterisk, located in the office, to another phone in the office. I'm wondering what possible sources there may be for this behavior? The big picture is, we are making out bound calls using AMI Originate commands. I have a script that does the originations using an AMI connection. Everything appears to be working well with this. It's clear that the Originate commands work, because the phones ring, and the dialplan/contexts work (except for WFS). We call over the Qwest lines using Originate, and we called locally (within the same office) using Originate. The only difference was that Qwest lines were not used in the latter. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
Thanks for the info. I didn't have any model in mind, just wondering what was required. Thanks again, much appreciated. Julian 2009/10/14 Jonathan Thurman jthurma...@gmail.com: Depends on what the router is. If you get a 2800 series router (we use 2801s and 2811s for T1s in production with no major issues). You need the T1/E1 module, DSPs, and an IOS that supports voice. For a 2800 series you would need something like: - VWIC2-MFT-T1/E1 ( or VWIC2-2MFT-T1/E1 if you want two ports) - PVDM2-32 (PVDM2-64 if you want two E1 ports, or two PVDM2-32s) - IOS that supports voice (I use spservicesk9) If you are looking at an older router like a 2651XM or something, you will need something like: - NM-HDV2-2T1/E1 (not the VWIC2, as the NM has build in ports) - PVDM2-32 If you have a specific router in mind, I can be more specific. -Jonathan On Wed, Oct 14, 2009 at 11:00 AM, Julian Lyndon-Smith aster...@dotr.com wrote: I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
The lowest end that you can use are 2600, 2600xm, 2800 or 3600. Then like a previous poster said, you need the DSP's and T1/E1 modules, but not all of them support it. NM-HDV2-2T1/E1 are relatively cheap, but you need to make sure that it actually has the t1/ei VWIC in it and it has DSP's in it as well. Some people sell the NM with nothing in it and that is useless. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, October 14, 2009 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco router Thanks for the info. I didn't have any model in mind, just wondering what was required. Thanks again, much appreciated. Julian 2009/10/14 Jonathan Thurman jthurma...@gmail.com: Depends on what the router is. If you get a 2800 series router (we use 2801s and 2811s for T1s in production with no major issues). You need the T1/E1 module, DSPs, and an IOS that supports voice. For a 2800 series you would need something like: - VWIC2-MFT-T1/E1 ( or VWIC2-2MFT-T1/E1 if you want two ports) - PVDM2-32 (PVDM2-64 if you want two E1 ports, or two PVDM2-32s) - IOS that supports voice (I use spservicesk9) If you are looking at an older router like a 2651XM or something, you will need something like: - NM-HDV2-2T1/E1 (not the VWIC2, as the NM has build in ports) - PVDM2-32 If you have a specific router in mind, I can be more specific. -Jonathan On Wed, Oct 14, 2009 at 11:00 AM, Julian Lyndon-Smith aster...@dotr.com wrote: I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ACD ASR
B.Masoud @ SH wrote: Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source destination? You asked this same question two weeks ago with the same subject. You got at least 5 answers. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Door Phones
Hi, Can anyone recommend a cheap SIP doorphone? Please only respond if you've had personal experience of a doorphone. Many thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
On Wed, Oct 14, 2009 at 12:57 PM, Julian Lyndon-Smith aster...@dotr.com wrote: Thanks for the info. I didn't have any model in mind, just wondering what was required. If you haven't purchased anything yet, or don't have anything, it might serve you better to look at other products. While the Cisco 2800s that we use work with Asterisk, we use them because that's what we had. I would look at an AudioCodes M1000, or an Adtran 908e or the like. I don't have any experience with E1, but I would guess that there is some support for them by those devices. The AudioCodes is about the same cost as a new Cisco solution, but the Adtran would probably be a lot less. I haven't had a chance to play with Adtran and Asterisk, but you can register at their website and play with all of the CLI / GUIs for all the devices which is really cool. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Phones
On Wed, Oct 14, 2009 at 1:52 PM, Dan Journo d...@keshercommunications.com wrote: Hi, Can anyone recommend a cheap SIP doorphone? Please only respond if you’ve had personal experience of a doorphone. I searched around for a while and couldn't find a hardened SIP external phone. We ended up using an ATA and a regular outside door phone from Ceeco (http://www.ceeco.net/ Model WPP-531-F). For a analog phone in a metal box, they aren't exactly cheap. You could say that an Analog phone would be more secure if someone ripped it off the wall, they wouldn't have network access. Then you just lock down what numbers can be called on your PBX. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Phones
Thanks for your email. I thought that might be the only real option. Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: 14 October 2009 22:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Door Phones On Wed, Oct 14, 2009 at 1:52 PM, Dan Journo d...@keshercommunications.com wrote: Hi, Can anyone recommend a cheap SIP doorphone? Please only respond if you've had personal experience of a doorphone. I searched around for a while and couldn't find a hardened SIP external phone. We ended up using an ATA and a regular outside door phone from Ceeco (http://www.ceeco.net/ Model WPP-531-F). For a analog phone in a metal box, they aren't exactly cheap. You could say that an Analog phone would be more secure if someone ripped it off the wall, they wouldn't have network access. Then you just lock down what numbers can be called on your PBX. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension Paging
Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Phones
You could take a look at the Valcom VIP-172L http://www.valcom.com/techsupport/ip_solutions/pagepro/ipdoorphones_web.pdf Or Cyberdata http://www.cyberdata.net/products/voip/digitalanalog/intercom/index.html Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: Wednesday, October 14, 2009 5:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Door Phones On Wed, Oct 14, 2009 at 1:52 PM, Dan Journo d...@keshercommunications.com wrote: Hi, Can anyone recommend a cheap SIP doorphone? Please only respond if you've had personal experience of a doorphone. I searched around for a while and couldn't find a hardened SIP external phone. We ended up using an ATA and a regular outside door phone from Ceeco (http://www.ceeco.net/ Model WPP-531-F). For a analog phone in a metal box, they aren't exactly cheap. You could say that an Analog phone would be more secure if someone ripped it off the wall, they wouldn't have network access. Then you just lock down what numbers can be called on your PBX. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Phones
On Wed, 2009-10-14 at 14:05 -0700, Jonathan Thurman wrote: On Wed, Oct 14, 2009 at 1:52 PM, Dan Journo d...@keshercommunications.com wrote: Hi, Can anyone recommend a cheap SIP doorphone? Please only respond if you’ve had personal experience of a doorphone. I searched around for a while and couldn't find a hardened SIP external phone. We ended up using an ATA and a regular outside door phone from Ceeco (http://www.ceeco.net/ Model WPP-531-F). For a analog phone in a metal box, they aren't exactly cheap. You could say that an Analog phone would be more secure if someone ripped it off the wall, they wouldn't have network access. Then you just lock down what numbers can be called on your PBX. snip We've just installed a CyberData VoIP intercom and are quite happy with it so far: http://www.cyberdata.net/products/voip/digitalanalog/intercom/index.html -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension Paging
Check on voip-info.org -Original Message-From: d...@keshercommunications.comSent: Wed, 14 Oct 2009 22:24:41 +0100To: asterisk-users@lists.digium.comSubject: [asterisk-users] Extension Paging Hi, We have SPA921 handsets which apparently support Paging, however i can’t find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension Paging
Already checked there, however it doesn’t really give a great deal of information. Many other links? Thanks Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Godbout Sent: 14 October 2009 22:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Extension Paging Check on voip-info.org -Original Message- From: d...@keshercommunications.com Sent: Wed, 14 Oct 2009 22:24:41 +0100 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Extension Paging Hi, We have SPA921 handsets which apparently support Paging, however i can’t find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension Paging
Try this link: http://tinyurl.com/yl7ra6w On Wed, Oct 14, 2009 at 6:26 PM, Dan Journo d...@keshercommunications.com wrote: Already checked there, however it doesn’t really give a great deal of information. Many other links? Thanks Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Godbout Sent: 14 October 2009 22:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Extension Paging Check on voip-info.org -Original Message- From: d...@keshercommunications.com Sent: Wed, 14 Oct 2009 22:24:41 +0100 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Extension Paging Hi, We have SPA921 handsets which apparently support Paging, however i can’t find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing sip calls work; incoming calls fail
On Wed, 14 Oct 2009 11:51:02 +0200 Ivan Stepaniuk i...@albafotonica.com wrote: listm...@websage.ca wrote: On Sat, 10 Oct 2009 18:02:04 -0700 listm...@websage.ca wrote: On Sun, 11 Oct 2009 02:11:47 +0200 Ivan Stepaniuk i...@albafotonica.com wrote: listm...@websage.ca wrote: On the LAN side I can see the INVITE and OKAY messages which end with a CANCEL, apparently initiated by the Asterisk gateway. On the WAN side I can see that my Asterisk gateway is repeatedly sending OKAY messages in response to the INVITE from my ITSP. I assume the trouble is that these messages are either not getting back to my provider or something is blocking the confirmation from them. This more or less confirms what was seen in the sip debug trace as well. Post that SIP message from the CLI (sip debug), try adding externip=XXX.XXX.XXX.XXX (your external/public IP address) to your sip.conf global section, asterisk may be including it's private address in the OKAY sent to your provider. Here's the last message in sip debug before it gives up: ... Retransmitting #6 (no NAT) to 66.51.127.173:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bKee03.6a63373.0;received=66.51.127.173 Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKpae7KcSeB2Bte Record-Route: sip:66.51.127.173;lr;ftag=9Z5N4eayXp3Qm From: 2508864577 sip:2508864...@66.51.127.163;tag=9Z5N4eayXp3Qm To: sip:12504129...@66.51.127.173;tag=as32af6364 Call-ID: b734bd26-2fae-122d-91a5-653b331e358a CSeq: 121440337 INVITE User-Agent: Asterisk PBX 1.6.0.15 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: sip:12504129...@96.50.76.138 Content-Type: application/sdp Content-Length: 262 v=0 o=root 992672626 992672626 IN IP4 96.50.76.138 s=Asterisk PBX 1.6.0.15 c=IN IP4 96.50.76.138 t=0 0 m=audio 15550 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical packet (see doc/sip-retransmit.txt). Scheduling destruction of SIP dialog '4e8ef1b977bf0e062212334634080...@192.168.11.1' in 6400 ms (Method: INVITE) ... 66.51.127.173 is my provider's SIP server 66.51.127.163 is my provider's RTP server I even check DNS to make sure both forward and reverse records jive. Externip was a good suggestion, and worth a try, though because I'm registering with my provider and using dynamic=yes, wouldn't they just reply to that anyway, especially given that the registration works fine? Anyway, after adding externip=my-external-ip to [general] and doing a sip reload in the console the problem remains... [Bumping this in the hope that someone might have some new insight or suggestions since I posted this on a holiday weekend (in my part of the world anyway)...] I was staring at the SIP transcript and I don't see anything wrong, I'm out of suggestions, except that you could analyze and compare the packets when your phone is connected directly (if it's physically possible). I hope someone throws some light over this. Ivan, Thanks for the sympathetic words. After studying the configuration until I was ready to scream and testing everything I could think of I came to the conclusion that it was not likely my very simple setup that was to blame. I opened a ticket with my ITSP and although they were initially quite eager to help, we ultimately couldn't sort it out. I figure they think it's all my fault shrug. I moved to another provider with nearly the same configuration and interestingly enough, the problem disappeared. Wish I knew why it failed in the first place but on the other hand I'm happy that I at least have a working phone system once again and can get back to my regular job. Thanks again for your interest in trying to help. GM -- Greg Maruszeczka ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI for multiple voice mail boxes
Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callpickup works for outside calls but not inside calls
Hello, all. I've got a problem where we set up call pickup for a
customer. If the Bob's extension rings and Bob is in Jim's office, Bob
can press the button on his Snom 320 that says Bob and pick up his
line. It works great for calls coming in from the outside but does not
work for internal calls. Internal calls generate a
app_directed_pickup.c:204 pickup_exec: No target channel found for 617
error.
I see an old bug about this where the contexts were not consistent but
ours appear to be consistent. Here are examples of pertinent parts of
the dialplan:
[a10base]
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
; Terry Keeley
; We put these in a10base rather than a10 or a10pub
;so that the spare stations can access them but public cannot
exten = 612,hint,SIP/tkeeley
; Joe Intrabartola
exten = 613,hint,SIP/jintrabartola
; Maryann Lapolla
exten = 614,hint,SIP/mlapolla
; Michael Intrabartola
exten = 616,hint,SIP/mintrabartola
; Vinny De Marco
exten = 617,hint,SIP/vdemarco
; Reception - the Reception desk may ring when someone dials zero
exten = 621,hint,SIP/reception-a10
; Steve McClain
exten = 624,hint,SIP/smcclain
; Amityville Intercom
;exten = 686,1,Dial(SIP/avilleextdoor-a10,60)
;exten = 686,n,Hangup()
exten = _*8XXX,1,Pickup(${EXTEN:2...@a10pub) ; Enable call pickup for hinted
stations
exten = 7998,1,VoiceMailMain(${CALLERID(num)}...@a10) ; Direct mail retrieval
exten = 7998,n,Hangup()
include = a10pub
include = a10utils
include = a10conf
include = a10parking
[a10in] ; direct inbound SIP dialing
exten = conference,1,Goto(a10pub,6000,1)
exten = joe,1,Goto(a10pub,613,1)
exten = maryann,1,Goto(a10pub,614,1)
exten = michael,1,Goto(a10pub,616,1)
exten = terry,1,Goto(a10pub,612,1)
exten = tommyvan,1,Goto(a10pub,615,1)
exten = vinny,1,Goto(a10pub,617,1)
exten = ebc,1,Goto(a10pub,9,ringall)
exten = vmail,1,Goto(a10pub,7999,1)
[a10pub]
; Public access - BE SURE there is no outbound access from here, e.g.,
; Background() functions will jump to any valid extension entered
; whether or not it is listed in the menu
; Terry Keeley
exten = 612,1,Set(__VM=612) ; VoiceMail ID
exten = 612,n,Gosub(a10ringtones,internal,1)
exten = 612,n,Macro(common,SIP/tkeeley,1,a10)
; 1 for VM, a10 VM context, no followme, ring for default seconds
exten = 8612,1,VoiceMail(6...@a10,u)
exten = 7612,1,VoiceMailMain(6...@a10)
exten = 7612,n,Hangup()
; Joe Intrabartola
exten = 613,1,Set(__VM=613)
exten = 613,n,Gosub(a10ringtones,internal,1)
exten = 613,n,Macro(common,SIP/jintrabartola,1,a10)
exten = 8613,1,VoiceMail(6...@a10,u)
exten = 7613,1,VoiceMailMain(6...@a10)
; Vinny De Marco
exten = 617,1,Set(__VM=617)
exten = 617,n,Gosub(a10ringtones,internal,1)
exten = 617,n,Macro(common,SIP/vdemarco,1,a10)
exten = 8617,1,VoiceMail(6...@a10,u)
exten = 7617,1,VoiceMailMain(6...@a10)
; Floral Park Spare
exten = 618,1,Gosub(a10ringtones,internal,1)
exten = 618,n,Dial(SIP/sparef1-a10,120,o) ; Ring the phone for up to 2 minutes
exten = 618,n,Hangup()
If I make a SIP call across the Internet to Vinny, for example, we issue
a goto to Vinny's internal extension. Terry can press the call pickup
and it all works. The same if I dial in from the PSTN. Here is the
call sequence:
-- Executing [vi...@a10in:1] Goto(SIP/jasiii-ad0e1048, a10pub,617,1) in
new stack
-- Goto (a10pub,617,1)
-- Executing [...@a10pub:1] Set(SIP/jasiii-ad0e1048, __VM=617) in new
stack
-- Executing [...@a10pub:2] Gosub(SIP/jasiii-ad0e1048,
a10ringtones,internal,1) in new stack
-- Executing [inter...@a10ringtones:1] SIPAddHeader(SIP/jasiii-ad0e1048,
Alert-Info: http://www.notused.com\;info=alert-internal\;x-line-id=0) in
new stack
-- Executing [inter...@a10ringtones:2] Return(SIP/jasiii-ad0e1048, ) in
new stack
-- Executing [...@a10pub:3] Macro(SIP/jasiii-ad0e1048,
common,SIP/vdemarco,1,a10) in new stack
-- Executing [...@macro-common:1] Set(SIP/jasiii-ad0e1048, TM=24) in
new stack
-- Executing [...@macro-common:2] Dial(SIP/jasiii-ad0e1048,
SIP/vdemarco,24,o) in new stack
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
-- Called vdemarco
-- SIP/vdemarco-d4012df8 is ringing
-- SIP/vdemarco-d4012df8 is ringing
-- SIP/vdemarco-d4012df8 is ringing
-- SIP/vdemarco-d4012df8 is ringing
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
== Extension Changed 612[a10base] new state InUse for Notify User
jintrabartola
== Extension Changed 612[a10base] new state InUse for Notify User
reception-a10
-- Executing [*8...@a10f:1] Pickup(SIP/tkeeley-acc9aaf8, 6...@a10pub)
in new stack
== Extension Changed 612[a10base] new state InUse for Notify User
mintrabartola
== Extension Changed 612[a10base] new state InUse for Notify User mlapolla
== Extension Changed 612[a10base] new state InUse for Notify
