[asterisk-users] AMI - Getting Event of QueueAgents WrapupTime State
Hi @ all, in the reason on having some agents logged in in more than one queues I need to get the state if an Agent goes in Postprocessing State to do this for this Agent in all other Queues he is logged on. For this I tried to catch this Event above the AMI. But there is never thrown the QueueMemberPaused or AgentComplete Event and in all the QueueMemberStatus Events the Agent never has the state Paused set. What options do I have to to solve this? Best regards Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cell Phone as a Queue member
Hi, Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries to call SIP/$TRUNK instead. Cheers, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [NAT] SSH vs. OpenVPN?
Hello In case a NAT firewall prevents using STUN to open SIP/RTP ports, a solution is to first connect the phone to the Asterisk server through a tunnel, and then have data go through the tunnel. Are there hardphones that support OpenVPN? If none, what about SSH? Is this a good alternative to use VoIP with SIP? If you've tried either or both solutions, I'm interested in any feedback. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
Gilles wrote: Are there hardphones that support OpenVPN? I've seen people mention snom with OpenVPN: http://wiki.snom.com/Networking/Virtual_Private_Network_%28VPN%29 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
Hi Gilles, You can't tunnel UDP through SSH. Some of the newer Grandstream handsets support OpenVPN and are a bit cheaper than the Snom alternatives. - Regards, AJ Stanfield t: 0161-850-4001 e: a...@dmcip.com w: http://www.dmcip.com - Original Message - From: Gilles codecompl...@free.fr To: asterisk-users@lists.digium.com Sent: Tuesday, 31 January, 2012 12:32:20 PM Subject: [asterisk-users] [NAT] SSH vs. OpenVPN? Hello In case a NAT firewall prevents using STUN to open SIP/RTP ports, a solution is to first connect the phone to the Asterisk server through a tunnel, and then have data go through the tunnel. Are there hardphones that support OpenVPN? If none, what about SSH? Is this a good alternative to use VoIP with SIP? If you've tried either or both solutions, I'm interested in any feedback. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
hello, yeallink T26 and T28 support OpenVPN too Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 12:54:41 + (GMT), Arthur Stanfield a...@dmcip.com wrote: You can't tunnel UDP through SSH. Some of the newer Grandstream handsets support OpenVPN and are a bit cheaper than the Snom alternatives. Thanks for the infos. So the only way to use SIP through locked-down NAT routers is to use OpenVPN, either with the few hardphones that support it or with a softphone on a computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 07:57:22 -0500, bakko asannu...@gmail.com wrote: yeallink T26 and T28 support OpenVPN too Thanks for the infos. If someone tried the Snom, Grandstream, or Yeallink, how good is their OpenVPN connection? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CA Issued Certificates / TLS + SRTP
Hi Daniel, Thank you very much for your responses! At least I only wasted 5 hours on the chained certificate issue. I have some responses / questions below. The certificate is a GeoTrust Rapid SSL certificate. I have received the my server specific crt file and also an intermediate certificate. Intermediate certificates work for some user agents (e.g. my Polycom). There has been speculation that they won't work with some older UAs Ultimately, most of the budget priced certificates are signed with an intermediate cert, and OpenSSL supports it, so there is no reason Asterisk shouldn't support this. You asked a question as to what people have experience with. When I googled, the only response I found was this one which said Comodo didn't work with Microsoft: http://pbxinaflash.com/forum/showthread.php?t=11001 I quickly did a search using SSL shopper when I wanted to purchase a real certificate and they said all 8 certificates they had on record for a single domain were chained. I think this is a new requirement of 256 bit encryption so as you pointed out (and if I read the Rapid SSL page properly), we aren't going to get away from it. Yes, in the correct order Currently, Asterisk expects the key and cert together in the same file: I think that is bad, but that is the way it is: https://issues.asterisk.org/jira/browse/ASTERISK-19267 I will give this a shot later on tonight... * And, is it necessary to use both my server specific certificate and the intermediate certificate on the telephones or will the telephones only require the server specific certificate? The phones should already have the root certificate for Geotrust, you should not deploy intermediate roots into the phones if you can avoid it If I understand this correctly (and the other emails you sent), the Polycom does not need any preloaded certificates / keys, it will ask the CA and then evaluate the certificate provided by Asterisk during TLS setup; is that correct? Kind Regards Stuart -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DTMF in Voicemail main
On 01/31/2012 12:17 AM, Ira wrote: Tonight I tried 4 versions of Asterisk; 10.0.0, 10.0.1, 10.1.0 and trunk. On 10.1.0 and trunk, I can't successfully enter the password for any mailbox in voicemailmain on my Aastra 480i phones. All four version work with a Snom cordless SIP phone. In 10.0.0 and 10.0.1 the Aastra works perfectly. So needless to say I'm back to running 10.0.1. The WAF is very low for stuff like that. It is quite unlikely that there were any changes between 10.0.1 and 10.1.0 that would affect DTMF detection or app_voicemail itself, but it's certainly possible. That's why we have an issue reporting system, and it's also why we produce release candidates to get testing prior to making official releases. I notice that comedian mail has instead of [] brackets. Does that mean it's on its way to being deprecated? I assume you are referring to how app_voicemail (not 'comedian mail') is listed the menuselect tool. Umm... no, those are completely unrelated. How did you reach that assumption? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell Phone as a Queue member
Use local channel 2012/1/31 Niccolò Belli darkbas...@gmail.com: Hi, Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries to call SIP/$TRUNK instead. Cheers, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell Phone as a Queue member
C F wrote: Use local channel You'll also want to keep track of the number of active calls, since, I believe, the queue app will not be able to see signaling on that line. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell Phone as a Queue member
Alternately, you could use a SIP channel with followme or the newer releases have some Bluetooth capabilities. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, January 31, 2012 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cell Phone as a Queue member C F wrote: Use local channel You'll also want to keep track of the number of active calls, since, I believe, the queue app will not be able to see signaling on that line. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRV record for non-standard SIP port?
Hello To cut down on the number of hackers trying to break into an Asterisk server, I'd like to simply move the SIP port from the standard UDP 5060 to something non-standard. Since this server must be able to receive INVITEs from any SIP UA (server or client), it appears that I must add an SRV record in the DNS so that they can locate the server and the port used to reach it. _sip._udp SRV 0 5060 host.tld. www.voip-info.org/wiki/view/DNS+SRV Are there pitfalls/traps I must pay attention to before going ahead and add that type of record in the DNS? What about internal SIP clients that register with Asterisk: Will they query the DNS to find the SIP port also, or must reconfigure them all to use the non-standard port Asterisk listens on? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
On Tue, 2012-01-31 at 14:13 +0100, Gilles wrote: On Tue, 31 Jan 2012 07:57:22 -0500, bakko asannu...@gmail.com wrote: yeallink T26 and T28 support OpenVPN too Thanks for the infos. If someone tried the Snom, Grandstream, or Yeallink, how good is their OpenVPN connection? Using Yealink T-28 with OpenVPN works fine - about three weeks now with no issues. Bummed that it seems to only support one tunnel, though. I asked their support team if they could make whatever changes necessary to support multiple, and their response made it sound promising :) I love this phone, actually. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 10:03:46 -0600, Jeff LaCoursiere j...@sunfone.com wrote: Using Yealink T-28 with OpenVPN works fine - about three weeks now with no issues. Bummed that it seems to only support one tunnel, though. I asked their support team if they could make whatever changes necessary to support multiple, and their response made it sound promising :) Thanks for the feedback. Multiple tunnels are for conference calls? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
Jeff LaCoursiere wrote: Bummed that it seems to only support one tunnel, though As in you can't register the phone to more then 1 remote Asterisk server via 2 different VPN tunnels or you can't have more then 1 call per VPN link? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
On Tue, 2012-01-31 at 11:29 -0500, Doug Lytle wrote: Jeff LaCoursiere wrote: Bummed that it seems to only support one tunnel, though As in you can't register the phone to more then 1 remote Asterisk server via 2 different VPN tunnels or you can't have more then 1 call per VPN link? The former - I have the phone registered to several asterisk servers, and would like to have multiple tunnels in place to each of those asterisk servers, which it will not do. Multiple calls through the tunnel is no problem. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
On Tue, 2012-01-31 at 17:23 +0100, Gilles wrote: On Tue, 31 Jan 2012 10:03:46 -0600, Jeff LaCoursiere j...@sunfone.com wrote: Using Yealink T-28 with OpenVPN works fine - about three weeks now with no issues. Bummed that it seems to only support one tunnel, though. I asked their support team if they could make whatever changes necessary to support multiple, and their response made it sound promising :) Thanks for the feedback. Multiple tunnels are for conference calls? No - the phone allows you to register with multiple servers, and I would like to reach each server over its own tunnel. It won't do that today. I've had good luck requesting features from Yealink and having them show up in new firmware releases, though, and I think this will probably go that way too. OpenVPN supports multiple tunnels, and I am not sure how they managed to break it in such a way that it won't on their platform. To make it work you have to name the tunnel conf file vpn.conf, and I am sure their openvpn startup routines are just hardcoded for that one conf file. I don't expect it would be a major mod to look for additional conf files at startup, like the stock init.d scripts do... Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 10:44:12 -0600, Jeff LaCoursiere j...@sunfone.com wrote: No - the phone allows you to register with multiple servers, and I would like to reach each server over its own tunnel. It won't do that today. Thanks for the info. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRV record for non-standard SIP port?
On 31/01/12 16:16, Gilles wrote: Hello To cut down on the number of hackers trying to break into an Asterisk server, I'd like to simply move the SIP port from the standard UDP 5060 to something non-standard. Something more appropriate for your goal might be a move to TLS, it is definitely needed for any external connectivity This RFC provides some details: http://tools.ietf.org/html/rfc5922 The bottom line is that external SIP peers must send you their cert when they connect. SIP hackers will need to identify themselves (e.g. with credit card) to get a certificate, or they just won't be able to talk to your server. Obviously, this cuts out about 99% of the script kiddies. As a further safety measure, you could use something like repro or Kamailio as a SIP router to isolate your Asterisk from the public internet. All DNS SRV records would point at the SIP router, not Asterisk. Phones would register with the SIP router. Calls would be selectively routed to Asterisk (e.g. for voicemail) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Proposed changes to Asterisk release and support cycles
I've created a page on wiki.asterisk.org outlining some changes we're proposing to make to the Asterisk release and support cycles. As always, before implementing any changes of this type, we'd like to collect some community feedback on the proposal. The page is here: https://wiki.asterisk.org/wiki/x/5ggiAQ Feel free to comment here, or on the page itself if you find any errors or inconsistencies in the page's content. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Deadlock detected in asterisk-1.8.9.0 x86_64
I am having problems with a deadlock in Asterisk 1.8.9.0. The system is a x86_64 machine that is being used as a callcenter. The agents log in via the AgentLogin application, and each Agent/ channel is assigned to one or more queues. A custom separate process generates calls into the queues for the agents to answer. The calls all go out through a SIP trunk, and all of the agent extensions are SIP. After an hour or so, asterisk deadlocks. Any attempt to run agent show or agent show online through the console hangs. Also, AMI events seem to stop. However, the users seem to be still connected, only they do not receive calls anymore (the custom process waits forever for the Originate response). The deadlock is apparently spontaneous - there is no explicit action taken by the administrator that seems to induce the issue. I will try to make sense of the attached traces, but I hope someone on the list could provide a clue on what to look for. Backtraces attached to https://issues.asterisk.org/jira/browse/ASTERISK-19285 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
From my perspective this makes a lot more sense than the current cycle. My big issue is with patches that have new features. Not having them in a trunk released version adds a lot of issues trying to support them in house. I like the idea of LTR release more often that would have the feature patches baked in. Case in point the new conference app requires a jump to version 10 while the 1.8 conference app is quite useless but 1.8 is my LTR version so I am stuck without the conference app in my mainline systems for two years. This new method would reduce the time for situations like this. This is the same with the F option in faxReceive as well. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Kevin P. Fleming kpflem...@digium.com Sent: Tuesday, January 31, 2012 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Proposed changes to Asterisk release and support cycles I've created a page on wiki.asterisk.org outlining some changes we're proposing to make to the Asterisk release and support cycles. As always, before implementing any changes of this type, we'd like to collect some community feedback on the proposal. The page is here: https://wiki.asterisk.org/wiki/x/5ggiAQ Feel free to comment here, or on the page itself if you find any errors or inconsistencies in the page's content. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Troubleshooting one-way audio with H.323 trunk between Asterisk and Avaya IP Office
I'm attempting to configure an H.323 trunk (using chan_h323) between an Asterisk box and an Avaya IP office. It mostly works. Calls from Polycom SIP devices registered to Asterisk can place calls over the trunk to IP Office extensions and everything works great. However, calling from an IP Office handset to any of the Polycoms results in a one-way call where the Polycom can not hear the Avaya. I think what's happening is somehow, the Polycom is receiving two RTP streams, and one of them is silence. I think this because if I place the call on hold, I will hear, occasionally, short bursts of what could be the hold music on the Polycom. Also, when I look at packet captures taken from the Polycom's port, I see two streams when it's working (the Polycom calls Avaya) and three when it's not (Avaya calls Polycom). I do notice that this extra stream has a unique SSRC. If I do the packet captures from the Asterisk box, it looks like the extra stream is generated by Asterisk. Graphically, the RTP streams and their SSRCs I see on a working call (Polycom calls Avaya) looks like this: Polycom --0x123- Asterisk --0x123- IP Office -0x456-- -0x456 The non-working call (Avaya calls Polycom) looks like: -0x789-- Polycom --0x123- Asterisk --0x123- IP Office -0x456-- -0x456 However, I'm new to Asterisk, and I'm not very familiar with any of these VoIP protocols, so I find myself stuck. Can anyone suggest some troubleshooting steps or material I might read which would help me to determine if my hypothesis is correct, and if so, how I can determine what's responsible for this extra stream? Console output from a non-working call with sip and h323 debug and trace on follows. Here, 207 is the Avaya handset originating the call, and 216 is the Polycom receiving it. 172.20.20.233 is the IP Office, 172.20.20.205 is Asterisk, and 172.20.32.70 is the Polycom. asterisk01*CLI == New H.323 Connection created. --Received SETUP message -- Setting up Call -- Call token: [ip$172.20.20.233:4232/71] -- Calling party name: [] -- Calling party number: [207] -- Called party name: [216] -- Called party number: [216] -- Calling party IP: [172.20.20.233] Setting capabilities to 0x4 (ulaw) Capabilities in preference order is (ulaw) DTMF mode is 8 Allowed Codecs for ip$172.20.20.233:4232/71 (ip$172.20.20.205:1720): Table: G.711-uLaw-64k 1 UserInput/hookflash 2 UserInput/basicString 3 Set: 0: 0: G.711-uLaw-64k 1 1: UserInput/hookflash 2 2: UserInput/basicString 3 =-= In OnAnswerCall for call 71 - Progress Indicator: 0 - Inserting PI of 8 into ALERTING message -- Executing [216@default:1] Dial(H323/ip$172.20.20.233:4232/71, SIP/216) in new stack == Using SIP RTP CoS mark 5 Audio is at 172.20.20.205 port 10488 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 172.20.32.70:5060: INVITE sip:216@172.20.32.70 SIP/2.0 Via: SIP/2.0/UDP 172.20.20.205:5060;branch=z9hG4bK799c13df;rport Max-Forwards: 70 From: 207 sip:207@172.20.20.205;tag=as1de701d2 To: sip:216@172.20.32.70 Contact: sip:207@172.20.20.205 Call-ID: 4c0201703a430c12490761fe7d6cdead@172.20.20.205 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9-2+squeeze4 Date: Tue, 31 Jan 2012 19:41:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 247 v=0 o=root 992661417 992661417 IN IP4 172.20.20.205 s=Asterisk PBX 1.6.2.9-2+squeeze4 c=IN IP4 172.20.20.205 t=0 0 m=audio 10488 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 216 --- SIP read from UDP:172.20.32.70:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.20.20.205:5060;branch=z9hG4bK799c13df;rport From: 207 sip:207@172.20.20.205;tag=as1de701d2 To: sip:216@172.20.32.70;tag=9BF9794A-AD4778F1 CSeq: 102 INVITE Call-ID: 4c0201703a430c12490761fe7d6cdead@172.20.20.205 Contact: sip:216@172.20.32.70 User-Agent: PolycomSoundStationIP-SPIP_6000-UA/3.0.4.0061 Content-Length: 0 - --- (9 headers 0 lines) --- --- SIP read from UDP:172.20.32.70:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.20.20.205:5060;branch=z9hG4bK799c13df;rport From: 207 sip:207@172.20.20.205;tag=as1de701d2 To: sip:216@172.20.32.70;tag=9BF9794A-AD4778F1 CSeq: 102 INVITE Call-ID: 4c0201703a430c12490761fe7d6cdead@172.20.20.205 Contact: sip:216@172.20.32.70 User-Agent: PolycomSoundStationIP-SPIP_6000-UA/3.0.4.0061 Allow-Events: talk,hold,conference Content-Length: 0 - --- (10 headers 0 lines) --- -- SIP/216-0006 is ringing Sending alerting --- SIP read from UDP:172.20.32.70:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP
Re: [asterisk-users] Cell Phone as a Queue member
Il 31/01/2012 15:42, C F ha scritto: Use local channel Thanks, I completely forget about local channel. Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell Phone as a Queue member
Il 31/01/2012 15:46, Doug Lytle ha scritto: You'll also want to keep track of the number of active calls, since, I believe, the queue app will not be able to see signaling on that line. I'm sorry but, how to? Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
I like the idea of LTR release more often that would have the feature patches baked in. Case in point the new conference app requires a jump to version 10 while the 1.8 conference app is quite useless but 1.8 is my LTR version so I am stuck without the conference app in my mainline systems for two years. Well said! This is also true of any type of long term supported release whether if it's an operating system, application, etc. In the "LTS" name, it conjurs up thoughts of Ubuntu, but comparisons to RHEL/Fedora are far more appropriate I would think as Ubuntu focuses nearly exclusively on new point releases while backporting new features is what a company like Red Hat excels at and should be the prime example of how to run dual software channels (enterprise release in RHEL vs. hobby release in Fedora). Red Hat works so well for server systems because features are regularly backported with a *huge* emphasis on never breaking abi or build environments. So far there really hasn't been a lot of noticeable features backported to 1.8.x that I'm aware of, but then again 10 is the first release after 1.8. Generally, if there isn't a lot of support in maintaining a long term release, then it turns into merely a "old release that occasionally has quality security updates". This is a perfectly valid approach too, but so far Digium's use of "LTS" doesn't really clarify clearly to me which type they are meaning to confer: 1) release that will stay static for its entire release sans security updates or 2) release that will stay compatible throughout the software's life time while occasionally having features backported with development funded from paying clients with support contracts. It should also be said that the long term release really isn't the appropriate place to debut new technology. If you absolutely require the newest stuff that Digium produces, regardless of their LTS paradigm, the LTS release probably isn't meant for you. Using the RHEL/Fedora example of earlier, RHEL's backports only come through around once a year during the point releases. Anything more would be chaotic and against the notions of a long term supported release. Fedora gets new stuff every 6 months, freshly baked with some stuff just not working all that well. I know distros and applications are two fundamentally different things, with entirely different goals and requirements, but I still think Red Hat provides the best example because 1) they have turned it into a science how smooth their development process goes in ratio to satisfied customers and 2) it's the only other open source software project I can think of that can accurately compare. In a past meeting I had with Digium while working for another company, they too directly drew a correlation between the new LTS idea and ubuntu lts/non-lts and rhel/fedora. The conference app changes since 1.4 I haven't been thrilled with, but in the whole time I've been supporting 1.8.x for my customers, I've come up with a very stable solution building on it and I haven't had any surprises come my way. But think back before 1.8.x and Digium's plan for LTS: We lived in a world where 1.4 bounced back and forth between "ultra-stable" and "whoops, dtmf is completely borked again" largely due to the fact that a complete rewrite of various parts of Asterisk would greatly undermine projects written specifically for that branch so small fixes netted breakage in other parts of the software. And we also had 1.6.x which for 95% of stuff was brilliant, but that other 5% was so crucial that it delayed adoption. Personally, I don't think what Digium is doing is necessarily a perfect approach (hey, what is? we're all human), but they've vastly improved the quality of Asterisk from a support perspective. John Knight Classic City Telco LLC Email: j...@classiccitytelco.com | Main: (706) 995-0200 Direct: (706) 995-0201 | Mobile: (706) 255-9203 On 1/31/2012 2:20 PM, Bryant Zimmerman wrote: From my perspective this makes a lot more sense than the current cycle. My big issue is withpatches that have new features. Not having them in a trunk released version adds a lot of issues trying to support them in house. I like the idea of LTR release more often that would have the feature patches baked in. Case in point the new conference app requires a jump to version 10 while the 1.8 conference app is quite useless but 1.8 is my LTR version so I am stuck without the conference app in my mainline systems for two years. This new method would reduce the
[asterisk-users] Experience with Eicon Diva PRO 3.0?
Hello *, is here someone with an experience of the Eicon Diva PRO 3.0? I need 2 or 3 cards to connect an Anlagenanschluß (I do not know the name in englisch, but is 2 or more lines with the same numbers) to my intranet servers (IBM x335/x345 - requires PCI-X or at least PCI 2.1). Thanks, Greetings and nice Day/Evening Michelle Konzack -- # Debian GNU/Linux Consultant ## Development of Intranet and Embedded Systems with Debian GNU/Linux Internet Service Provider, Cloud Computing http://www.itsystems.tamay-dogan.net/ itsystems@tdnet Jabber linux4miche...@jabber.ccc.de Owner Michelle Konzack Gewerbe Strasse 3 Tel office: +49-176-86004575 77694 Kehl Tel mobil: +49-177-9351947 Germany Tel mobil: +33-6-61925193 (France) USt-ID: DE 278 049 239 Linux-User #280138 with the Linux Counter, http://counter.li.org/ signature.pgp Description: Digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experience with Eicon Diva PRO 3.0?
On 31-01-12 22:47, Michelle Konzack wrote: Hello *, is here someone with an experience of the Eicon Diva PRO 3.0? I need 2 or 3 cards to connect an Anlagenanschluß (I do not know the name in englisch, but is 2 or more lines with the same numbers) to my intranet servers (IBM x335/x345 - requires PCI-X or at least PCI 2.1). If that is an Eicon Diva *Server* card then it should work fine with the drivers and chan_capi from http://www.melware.de Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
Why the short life on Asterisk 10? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, January 31, 2012 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Proposed changes to Asterisk release and support cycles I've created a page on wiki.asterisk.org outlining some changes we're proposing to make to the Asterisk release and support cycles. As always, before implementing any changes of this type, we'd like to collect some community feedback on the proposal. The page is here: https://wiki.asterisk.org/wiki/x/5ggiAQ Feel free to comment here, or on the page itself if you find any errors or inconsistencies in the page's content. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
On 01/31/2012 06:57 AM, Gilles wrote: Thanks for the infos. So the only way to use SIP through locked-down NAT routers is to use OpenVPN, either with the few hardphones that support it or with a softphone on a computer. You can also setup OpenVPN to connect a remote subnet (remote office) and it will route all traffic between subnets. Configure the hard/soft phones on the remote subnet to route through the OpenVPN. This works pretty well for me. -- The truth speaks for itself. I'm just the messenger. Lyta Alexander - Babylon 5 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
On 01/31/2012 04:07 PM, Danny Nicholas wrote: Why the short life on Asterisk 10? Can you rephrase your question? This proposal does not change the planned lifetime of Asterisk 10 at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
1.8 and 11 forward all seem to have a timeline of around 5 years. 10 only runs for two. Since the code is available that isn't a biggie to me, but the appearance of a short life could cause some customer discomfort. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, January 31, 2012 5:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Proposed changes to Asterisk release and support cycles On 01/31/2012 04:07 PM, Danny Nicholas wrote: Why the short life on Asterisk 10? Can you rephrase your question? This proposal does not change the planned lifetime of Asterisk 10 at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
On 01/31/2012 05:14 PM, Danny Nicholas wrote: 1.8 and 11 forward all seem to have a timeline of around 5 years. 10 only runs for two. Since the code is available that isn't a biggie to me, but the appearance of a short life could cause some customer discomfort. You must be looking at a different diagram than I am. Asterisk 1.8, 11, 13, 15 (and future) all have timelines of 4+1 years because they are LTS releases. Asterisk 10, 12, 14 (and future) have timelines of 1+1 years, because they are not LTS releases. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, January 31, 2012 5:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Proposed changes to Asterisk release and support cycles On 01/31/2012 04:07 PM, Danny Nicholas wrote: Why the short life on Asterisk 10? Can you rephrase your question? This proposal does not change the planned lifetime of Asterisk 10 at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
To properly phrase the question, why is 10.X not an LTS release? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, January 31, 2012 5:20 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Proposed changes to Asterisk release and support cycles On 01/31/2012 05:14 PM, Danny Nicholas wrote: 1.8 and 11 forward all seem to have a timeline of around 5 years. 10 only runs for two. Since the code is available that isn't a biggie to me, but the appearance of a short life could cause some customer discomfort. You must be looking at a different diagram than I am. Asterisk 1.8, 11, 13, 15 (and future) all have timelines of 4+1 years because they are LTS releases. Asterisk 10, 12, 14 (and future) have timelines of 1+1 years, because they are not LTS releases. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, January 31, 2012 5:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Proposed changes to Asterisk release and support cycles On 01/31/2012 04:07 PM, Danny Nicholas wrote: Why the short life on Asterisk 10? Can you rephrase your question? This proposal does not change the planned lifetime of Asterisk 10 at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
The LTR releases have a shorter support/life cycle and 10 is not an LTR. That accounts for the shorter life on 10. This is why I like this proposal. We would get faster LTR releases and that would allow us to have newer features sooner but still offer existing deployments security fixes on LTR releases. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
On 01/31/2012 05:20 PM, Danny Nicholas wrote: To properly phrase the question, why is 10.X not an LTS release? Umm... because it wasn't planned to be, and the decision was made nearly two years ago? Asterisk 1.8 is an LTS release, and now (with this proposal) Asterisk 11 would also be an LTS release. There's really no need for another LTS release sandwiched between them, and we can't commit Digium's development resources to extending the lifetime of that branch by three years. In addition, Asterisk 10 contains some significantly redesigned code that would not have been considered LTS-ready when Asterisk 10 was released. As we've said many times before, if a group of interested and committed community developers want to continue provide support for Asterisk 10 beyond its stated lifetime, we'll happily provide them the resources to be able to produce releases and make them available to the user community. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] SSH vs. OpenVPN?
On Tue, 31 Jan 2012 16:25:37 -0600, Dale Noll dn...@wi.rr.com wrote: You can also setup OpenVPN to connect a remote subnet (remote office) and it will route all traffic between subnets. Configure the hard/soft phones on the remote subnet to route through the OpenVPN. This works pretty well for me. Thanks for the info. I was thinking of connecting while on the road/vacation, but it's a good use to connect a remote office to the main office. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRV record for non-standard SIP port?
On Tue, 31 Jan 2012 18:22:41 +0100, Daniel Pocock dan...@readytechnology.co.uk wrote: Something more appropriate for your goal might be a move to TLS, it is definitely needed for any external connectivity [...] As a further safety measure, you could use something like repro or Kamailio as a SIP router to isolate your Asterisk from the public internet. Thanks for the tips. I'll read up on TLS and adding an SIP router in front of Asterisk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
I like it! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Congestion outbound only with ATA boxes
I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call out. It could not call out to the Link2VoIP or any of the SIP phones. I spent a lot of time going over the configureation for this Asterisk server and the settings in the Linksys PAP2T box but could not get it to work. I removed the Linksys PAP2T and replaced it with an HT503 because I read a lot of good recommendations for this device. It seems to have almost the same problem. I say almost because when the Linksys would get congestion I would hear the Asterisk recording tell me All circuits are busy now, good-bye but the HT503 only gets a busy tone. All the SIP phones can call out no problem but these two ATA boxes that I am trying to use the FXS ports to connect old analog POTS phones to are not working. I have turned on the debug in Asterisk and can see the point where I get congestion but I don't know how to make Asterisk give me more details as to why I am getting congestion. Can anyone help me to get more details about this problem? I traced the debug from a working SIP phone as it makes an outgoing call and from the HT503 as it tries to make a call. Everything is identical right up to the point where the HT503 gets a congestion instruction from the Asterisk server. Here is the debug output just at the point where it happens. -- AGI Script dialparties.agi completed, returning 0 -- Executing [s@macro-dial:7] Dial(SIP/302-08221a38, SIP/301||tr) in new stack -- Called 301 Home*CLI --- Transmitting (NAT) to 192.168.0.100:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.100:5060 ;branch=z9hG4bK443855200;received=192.168.0.100;rport=5060 From: sip:302@192.168.0.1;tag=1257222779 To: sip:301@192.168.0.1;tag=as201c8013 Call-ID: 979693319-5060-5@192.168.0.100 CSeq: 41 INVITE User-Agent: FPBX-2.4.0(1.4.19.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:301@192.168.0.1 Content-Length: 0 -- SIP/301-0822de30 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dial:8] Set(SIP/302-08221a38, DIALSTATUS=CONGESTION) in new stack -- Executing [s@macro-exten-vm:10] Set(SIP/302-08221a38, SV_DIALSTATUS=CONGESTION) in new stack -- Executing [s@macro-exten-vm:11] GosubIf(SIP/302-08221a38, 0?docfu|1) in new stack -- Executing [s@macro-exten-vm:12] GosubIf(SIP/302-08221a38, 0?docfb|1) in new stack -- Executing [s@macro-exten-vm:13] Set(SIP/302-08221a38, DIALSTATUS=CONGESTION) in new stack -- Executing [s@macro-exten-vm:14] NoOp(SIP/302-08221a38, Voicemail is novm) in new stack -- Executing [s@macro-exten-vm:15] GotoIf(SIP/302-08221a38, 1?s-CONGESTION|1) in new stack -- Goto (macro-exten-vm,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-exten-vm:1] PlayTones(SIP/302-08221a38, congestion) in new stack Audio is at 192.168.0.1 port 10162 Adding codec 0x100 (g729) to SDP -- Easy, fast GUI development. http://PerlQt.wikidot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
