[asterisk-users] Panic Button SMS Asterisk Integration
Has anyone done any integration of USB, etc. panic buttons and Asterisk? The basic idea would be to have a USB based panic button[1] along with a bit of code which would trigger a group SMS or perhaps a pre-recorded call to a group. Kind regards, Chris [1]http://www.amazon.com/StealthSwitch-Pro-USB-Foot-Pedal/dp/B00MI6K77K -- _ -- 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] Am I being hacked?
On Mon, Aug 19, 2013 at 2:40 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 08/19/2013 08:10 PM, Eric Wieling wrote:
One of Asterisk's dirty little secrets is that it does not show the
source IP when a device or hacker tries sending a call without registering.
The rejection message in the logs do not show the IP of the attacker.
Yes it sucks, yes it has been that way for many many years.
Are you aware of a patch that would show the source IP in the console and
logs?
I do something like this:
1. turn up the logging
2. add foo like this in my dial plan:
exten = _.,1,NoOp(Received incoming SIP connection from unknown peer to
${EXTEN})
exten = _.,n,Log(NOTICE,Anonymous peer IP: ${CHANNEL(peerip)})
exten = _.,n,Set(DID=${IF($[${EXTEN:1:2}=]?s:${EXTEN})})
exten = _.,n,Goto(s,1)
3. do some bar like this in my fail2ban filter:
VERBOSE.*SIP/HOST-.*Received incoming SIP connection from unknown peer
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice' (language '.*')
NOTICE.* .*: Anonymous peer IP: HOST
NOTICE.* .*: Failed to authenticate device .*\s?\sip:.*@HOST\.*
and that handles most of the hacking attempts I see on my system. I think
it may be possible for the second line to catch some false matches, but I
have not seen any issues with our system thus far.
Kind Regards,
Chris
PS. Feel free to comment on what is wrong with this and be sure to include
the right way to do it. :-)
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[asterisk-users] E911 Voip Trunking
During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if anyone else has attempted this. Kind Regards, Chris -- _ -- 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] E911 Voip Trunking
Section 6.5.2 (v4 interface) of NENA's v2 Interim Voip Architecture Standard shows a ladder diagram of their SIP flow which seems to match standard SIP. Maybe I'm oversimplifying it? [1] http://c.ymcdn.com/sites/www.nena.org/resource/collection/2851C951-69FF-40F0-A6B8-36A714CB085D/NENA_08-001-v2_Interim_VoIP_Architecture_i2.pdf On Fri, Apr 19, 2013 at 2:51 PM, Terry Brummell te...@brummell.net wrote: E911 does not follow the standard SIP RFC. That would be a good reason that they couldn't/wouldn't do it. Now that I say that I should qualify it and say NG911 (or ESINet) does not follow SIP RFC http://en.wikipedia.org/wiki/Next_Generation_9-1-1. That is not saying your county is not using standard SIP for E911, it just wouldn't be considered NG911. -- *From:* Chris Nighswonger *Sent:* Fri 4/19/2013 11:41 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] E911 Voip Trunking During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if anyone else has attempted this. Kind Regards, Chris -- _ -- 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] E911 Voip Trunking
On Fri, Apr 19, 2013 at 8:59 PM, Nathan Anderson nath...@fsr.com wrote: On Friday, April 19, 2013 5:35 PM, Warren Selby wrote: There are E911 providers that offer this functionality. I know off the top of my head, 911Enable offers a service like this. A former client of mine that provided hosted PBX services had a contract with them. I'm sure there are other providers out there as well. Indeed. 911ETC is who we use, and is another example. Even if you could peer directly with your county's PSAP, in the case of 911, I think it is a way better idea to go with one of these specialty SIP-based E911 providers, for the simple reason that even if you only sell VoIP service to people residing within your county Actually we are not reselling service and the majority of our phones are stationary. The few mobile soft phones we run would not need 911 service since they also carry cell phones, the soft phones being mainly remote extensions. So it sounds like it is at least worth pursuing. Kind Regards, Chris -- _ -- 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] Simple failover configuration
At present I have two hardware identically freepbx/asterisk boxes. The mysql db on one is slaved to the other and all config files are rsync'd once every 24 hours (we have few configuration changes). We use Polycom 321/331/550/650 phones, and I notice that these phones can be configured with two SIP servers. Would the simplest approach to failover be to just configure my primary asterisk server as the first SIP server and my backup as the second? Kind Regards, Chris -- _ -- 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] Call drop weirdness
On Wed, Oct 31, 2012 at 10:31 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. So I've been watching this problem and was finally able to get a pcap while it happened. snip Any thoughts on what might be going wrong? Do I need to post more info? Or am I on the wrong track altogether? After lots of grinding through traces and data dumps both on my end and my provider's, it turned out I was on the wrong track altogether. I finally threw together a script to log counter stats from the switchport into which our pbx is plugged, in spite of no noticeable counter activity. From this I found that the port was accumulating align errors at very slow rate; more like small bursts. So wrote a script to log this counter to an RRD and added it to a graph of traffic control rates. This allowed me to associate the bursts of align errors with RTP data flow. The graph here (http://www.screencast.com/t/vMsi3gVke4) contains two bursts of align errors. The first walked all over a call resulting in the outbound RTP stream dropping. As soon as the errors stopped the audio picked back up. The second burst correlates with log entries like this (no calls were placed or received during this burst): [2012-11-09 14:23:11] NOTICE[4199] chan_sip.c: Peer 'didforsale_outbound' is now UNREACHABLE! Last qualify: 84 [2012-11-09 14:23:13] NOTICE[4199] chan_sip.c: Peer 'didforsale_did' is now UNREACHABLE! Last qualify: 84 Interestingly enough, long ping sequences with large packet payloads do not seem to trigger any errors. Having changed cables, ports, as well as for duplex and speed mismatches, the only remaining hardware to be checked is the NIC, which I suspect is bad. So I'm going to switch over to our backup pbx and test that theory. I apologize for the lengthy explanation, but perhaps it will help some other person with a similarly maddening problem. Kind Regards, Chris -- _ -- 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] Fax Configuration
On Mon, Nov 5, 2012 at 10:43 PM, Vladimir Mikhelson v...@mikhelson.comwrote: My practical experience shows otherwise. I am able to receive faxes on SIP lines pretty reliably with no T.38 support. The biggest issue for me is CED tones detection. If CED is detected then fax reception goes on with no problems. I use this setup with both Asterisk receiving a fax and then e-mailing it to me as a PDF attachment This has been my experience also (with fax reception). Incoming faxes work for us nearly 100% of the time. I have not messed with fax sending and so cannot comment on that. Kind Regards, Chris -- _ -- 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] Call drop weirdness
I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. So I've been watching this problem and was finally able to get a pcap while it happened. I've attached a sanitized text version of the SIP signaling surrounding the time the outbound RTP stream dropped on this particular call. I'm no SIP expert, so there may not be enough info in the file to tell anything. A few notes about the file: 1. X.X.X.X is the public IP our asterisk server is behind. 2. Y.Y.Y.Y is the IP given to us by our provider to use in our SIP trunk through which inbound calls arrive. 3. Z.Z.Z.Z is the IP of our provider's server involved in the RTP stream. 4. DID is our DID. 5. CID is the number of the incoming caller. 6. The outbound RTP stream appears to drop three packets prior to the SIP BYE request. Any thoughts on what might be going wrong? Do I need to post more info? Or am I on the wrong track altogether? Kind Regards, Chris OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24883b1c;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as395fd02c To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 59e57daf68d31595480ca5927505485f@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24883b1c;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as395fd02c To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 59e57daf68d31595480ca5927505485f@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 SIP/2.0 503 Unable to load gateways Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24883b1c;rport=5060 From: Unknown sip:Unknown@X.X.X.X;tag=as395fd02c To: sip:Y.Y.Y.Y;tag=71fd1b189ab888f8d5fb24b00af87228.acb1 Call-ID: 59e57daf68d31595480ca5927505485f@X.X.X.X:5060 CSeq: 102 OPTIONS Server: DFSGW Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 SIP/2.0 503 Unable to load gateways Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport=5060 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y;tag=71fd1b189ab888f8d5fb24b00af87228.1bf8 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS Server: DFSGW Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK65801b0b;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as46c7a0b8 To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 0d78056a4c3c4b5d167b013c41450be9@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK65801b0b;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as46c7a0b8 To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 0d78056a4c3c4b5d167b013c41450be9@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:30 GMT Allow: INVITE,
[asterisk-users] Call drop weirdness
I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get a data dump to see what actually happens. Some times there is a bit of artifacting which takes place just prior to the drop, but mostly nothing: it just drops. I've checked and rechecked firewall settings. Bandwidth consumption on the Inet link varies, but the dropped audio happens even on off-peak times. I'm considering giving the Asterisk box a public IP on one IF and bypassing the FW to rule out NAT weirdness. Any thoughts on things to look at would be greatly appreciated. Kind Regards, Chris -- _ -- 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] WARNING T.30 ECM carrier not found
I'm working on setting up incoming fax reception on our * server. The majority of faxes come through fine. However each timed a fax comes in, I get a bunch of this: WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found Should this be of concern to me? A snip of the log is below. Kind Regards, Chris -- Executing [19108929322@from-trunk:6] Set(SIP/foobar_trunk_did_b-0174, CALLERPRES()=allowed_not_screened) in new stack -- Executing [19108929322@from-trunk:7] Set(SIP/foobar_trunk_did_b-0174, FAX_DEST=ext-fax^166^1) in new stack -- Executing [19108929322@from-trunk:8] Answer(SIP/foobar_trunk_did_b-0174, ) in new stack -- Executing [19108929322@from-trunk:9] Wait(SIP/foobar_trunk_did_b-0174, 4) in new stack == Redirecting 'SIP/foobar_trunk_did_b-0174' to fax extension due to CNG detection == Spawn extension (from-trunk, fax, 1) exited non-zero on 'SIP/foobar_trunk_did_b-0174' -- Executing [fax@from-trunk:1] Goto(SIP/foobar_trunk_did_b-0174, ext-fax,166,1) in new stack -- Goto (ext-fax,166,1) -- Executing [166@ext-fax:1] Set(SIP/foobar_trunk_did_b-0174, FAX_FOR=Fax (166)) in new stack -- Executing [166@ext-fax:2] NoOp(SIP/foobar_trunk_did_b-0174, Receiving Fax for: Fax (166), From: +18009806858 +18009806858) in new stack -- Executing [166@ext-fax:3] Set(SIP/foobar_trunk_did_b-0174, FAX_RX_EMAIL=f...@foobar.com) in new stack -- Executing [166@ext-fax:4] Goto(SIP/foobar_trunk_did_b-0174, s,receivefax) in new stack -- Goto (ext-fax,s,3) -- Executing [s@ext-fax:3] StopPlayTones(SIP/foobar_trunk_did_b-0174, ) in new stack -- Executing [s@ext-fax:4] ReceiveFAX(SIP/foobar_trunk_did_b-0174, /var/spool/asterisk/fax/1349791968.502.tif,f) in new stack -- Channel 'SIP/foobar_trunk_did_b-0174' receiving FAX '/var/spool/asterisk/fax/1349791968.502.tif' [2012-10-09 10:13:03] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:13:04] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:13:04] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:13:24] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:13:24] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:14:07] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:14:07] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:14:20] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:14:20] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found pbx1*CLI -- _ -- 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] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 4:37 AM, Michel Verbraak mic...@verbraak.org wrote: Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP portrange your asterisk will use for RTP traffic. change the rtpstart and rtpend to your needs and set them open in your FW. Do not make the range too small each active call will normally take one RTP channel incoming and one RTP channel outgoing. I have mine set to for example: rtpstart=1 and rtpend=10100. This should be enough for 100 simultanious calls. Thanks to everyone for the help in this regard. Its amazing how much I still do not know after nearly 30 years of wrestling with computers. :-) A lack of understanding about the nature of RTP led me to limit traffic inbound from specific IPs which, of course, led to inbound call weirdness. At this point I only have ~40 extensions, so I took Michel's advise and set my RTP range to 1-10100. The default 1 ports was a bit more surface area than I want to expose. Kind Regards, Chris -- _ -- 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] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 10:45 AM, Carlos Alvarez car...@televolve.com wrote: On Wed, Oct 3, 2012 at 7:35 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: At this point I only have ~40 extensions, so I took Michel's advise and set my RTP range to 1-10100. The default 1 ports was a bit more surface area than I want to expose. If you think 100 or 10k RTP ports going to your voice server makes ANY difference in security, you really need to re-think this and study more. Hi Carlos, I'm speaking of surface area. Ask any general if he would rather have to defend a 1000 mile front or a 1 mile front. You are right that an open port is an open port, but trying keeping the crowd out of 1 doors is *much* harder than trying to keep them out of 100 doors. Kind Regards, Chris -- _ -- 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] Call Termination Provider Madness
On Wed, Oct 3, 2012 at 12:09 PM, Carlos Alvarez car...@televolve.com wrote: And people, please stop trying to use human security to IP port analogies, they make you look foolish. -- Carlos Alvarez TelEvolve 602-889-3003 I stand corrected, Carlos. And thank-you for taking time to tell me how foolish I look. It is mostly true that we tend not to see our own foolishness and need to be told about it occasionally. Kind Regards, Chris -- _ -- 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] Call Termination Provider Madness
If this is the wrong place to post this I'm sure someone will let me know. :-) I'm looking for a reliable, inexpensive call termination service (SIP). The one I am presently with does not seem to know what IPs they send inbound calls from, and it is maddening to keep up with the FW changes necessary, not to mention the calls which are not connected. Kind regards, Chris -- _ -- 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] Call Termination Provider Madness
On Tue, Oct 2, 2012 at 5:30 PM, Chris Bagnall aster...@lists.minotaur.cc wrote: On 2/10/12 6:51 pm, Carlos Alvarez wrote: Your traffic level, number of concurrent calls, etc would help us know what sort of carrier you should be talking to. Equally important, your geographic location, and the geographic location to which most of your calls are made will be useful in helping list members advise you. We do ~4000+ min of outbound calling per month and just about that inbound. Not a large volume. We have four DID's (one of which is 800). Our calling patterns are mostly the lower 48 with a smattering international. We are located in NC. RTP is the problem in the FW. I just cannot see opening all RTP ports to $universal. But I'm probably missing a key piece of information. :-) Kind Regards, Chris -- _ -- 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] Remote SIP Extension Best Practices
What are best practices for allowing connection by remote SIP extensions over the internet? I'm thinking of putting the SIP inside a VPN connection. Kind Regards, Chris -- _ -- 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] Remote SIP Extension Best Practices
On Sat, Sep 29, 2012 at 12:41 PM, Carlos Rojas crt.ro...@gmail.com wrote: Hello. Vpn is good idea, is more secure, you can use tls with srtp as well. Are you using asterisk 1.8? Right? Asterisk 10.7.0 Kind Regards, Chris -- _ -- 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] Polycom Phone Configuration Overrides Not Saved
I have some Polycom 351 on Asterisk 10. On the same box as * I have a tftp server running to handle configs, etc. The Polycom phones have no problem grabbing config foo from the tftp server as well as writing log files back to the server. However, when I use the web-if on a phone to set a custom ring-tone, the web interface saves the change locally, but throws an error stating that it cannot write the override config to the tftp server. A look at a tcpdump shows that the phone indeed attempts to push the file to the server: 10:33:10.197294 IP 192.168.0.23.50672 pbx1.campus.foundations.edu.tftp: 38 RRQ 0004f2a5b892.cfg octet blksize 4096 10:33:10.473171 IP 192.168.0.23.29766 pbx1.campus.foundations.edu.tftp: 43 RRQ 2345-12365-001.sip.ld octet blksize 4096 10:33:10.711510 IP 192.168.0.23.33183 pbx1.campus.foundations.edu.tftp: 42 RRQ 0004f2a5b892_reg.cfg octet blksize 4096 10:33:10.758896 IP 192.168.0.23.28895 pbx1.campus.foundations.edu.tftp: 29 RRQ sip.cfg octet blksize 4096 10:33:11.620103 IP 192.168.0.23.21917 pbx1.campus.foundations.edu.tftp: 54 RRQ overrides/0004f2a5b892-phone.cfg octet blksize 4096 10:33:11.825803 IP 192.168.0.23.27460 pbx1.campus.foundations.edu.tftp: 52 RRQ overrides/0004f2a5b892-web.cfg octet blksize 4096 10:33:11.850646 IP 192.168.0.23.18554 pbx1.campus.foundations.edu.tftp: 55 RRQ licenses/-license.cfg octet blksize 4096 10:33:11.873077 IP 192.168.0.23.34766 pbx1.campus.foundations.edu.tftp: 55 RRQ licenses/0004f2a5b892-license.cfg octet blksize 4096 10:33:26.294928 IP 192.168.0.23.25504 pbx1.campus.foundations.edu.tftp: 37 WRQ cq_de_ku4dd.wav octet blksize 4096 10:33:36.238357 IP 192.168.0.23.63322 pbx1.campus.foundations.edu.tftp: 52 WRQ overrides/0004f2a5b892-web.cfg octet blksize 4096 10:33:36.539747 IP 192.168.0.23.37433 pbx1.campus.foundations.edu.tftp: 69 RRQ languages/Website_dictionary_language_en-us.xml octe A look at /var/log/messages shows: Sep 6 10:33:11 pbx1 in.tftpd[18368]: tftpd: read(ack): Connection refused Now why is it that the phone is refused only when writing the override file? Note that the only logging difference between a successful and unsuccessful write is the above line from the message log. The tcpdump looks the same. Kind Regards, Chris -- _ -- 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] Indicate multiple incoming calls from a multi-channel DID on a single phone
Is it possible to indicate multiple incoming calls from a multi-channel DID on a single phone? The phone in question is a Polycom 550. I've googled this with little to no success. Thanks, Chris -- _ -- 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] Indicate multiple incoming calls from a multi-channel DID on a single phone
On Mon, Sep 3, 2012 at 8:25 PM, Chris Nighswonger cnighswon...@foundations.edu wrote: Is it possible to indicate multiple incoming calls from a multi-channel DID on a single phone? The phone in question is a Polycom 550. I think I may have it, but would like some feedback so I won't chase the rabbit too far if it is something other than a rabbit... Set up 4 extensions, one assigned to each of the four line keys on the 550. Route the incoming call to line 1. Set up line n to roll over to line n+1 on busy. Am I going about this right? Thanks, Chris -- _ -- 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] Asterisk Package Question
Are there deb packages available for Asterisk 10 or for 11 beta? Kind Regards, Chris -- _ -- 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] VOIP over Metro Ethernet
I'm looking for any pros/cons of running an Asterisk based PBX over a metro ethernet pipe. The system will have about 40 handsets and 6 DIDs. Kind Regards, Chris -- _ -- 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] Cisco 30VIP Phone
Jason, Ok, the 30VIP template seems to be working ok as far as button assignment goes. I can define speeddial numbers to the speeddial buttons. However, it appears that there is no code to support the STIMULUS_SPEEDDIAL case. Is this correct? Debug output from chan_skinny shows the following when speeddial buttons are pressed on the 30VIP: Clearing Display unknown1 in handle_stimulus_message is '0' Received Stimulus: SpeedDial(1) unknown1 in handle_stimulus_message is '0' Received Stimulus: SpeedDial(2) unknown1 in handle_stimulus_message is '0' Received Stimulus: SpeedDial(3) Skinny [EMAIL PROTECTED] went on hook I'm still trying to come up to speed on the flow of code in chan_skinny.c Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
On 4/2/07, Jason Parker [EMAIL PROTECTED] wrote: Yes, you are (mostly) correct. Speeddials can be added to the phone, but they can't actually be used.. There is code there that is #ifdef'd out, because it (mostly) does not work. If this thread should be moved to the -dev list, just let me know. Bear with me as I am trying to get the broadest possible understanding of how this driver works. General questions: 1. Do I understand this relationship: *---chan_skinny---sccp-phone? 2. I have found a listing of SCCP request and response messages. Does chan_skinny simply translate these between * and the phone? 3. I notice some of the structs relating to the SCCP messages contain Unknown# elements. I assume these are data who's function is not yet understood. Specific questions regarding STIMULUS_SPEEDDIAL and the #ifdef code: 1. First we check to see if the button has a valid speeddial entry. 2. If it is, we try to grab a channel (line) (?)... 3. If we get a channel, we set a couple of variables (not sure about these yet)... 4. and setup some info about the state of the call (is this transmit to the phone or *? I think the phone...) 5. and clear the phone's display (It seems we do this alot.) 6. and give the phone dialtone (?)... 7. Then check with * to see if the number is valid (maybe in the dialplan) 8. Not sure what exactly happens here with the If not but if it evaluates, I think we have * place the call. 9. Break. I assume that functions beginning with 'ast_' belong to *. I'll stop here for this round. I understand that this may be a trivial item to fix, but it will go a long way toward familiarizing me with the code. Maybe then I'll be able to contribute to some of the more difficult issues. Thanks again, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
On 3/29/07, Jason Parker [EMAIL PROTECTED] wrote: - Chris Nighswonger [EMAIL PROTECTED] wrote: That is the conclusion I came to and was confirmed today in a very brief chat with one of the individuals listed as a developer on the chan_skinny module. He said that they could be implemented. What I would like to know, and do not understand, is the relationship between the code in chan_skinny.c which sets up the softkeys which are implimented and the actual key positions on the phone. With this info, I can hack the code to impliment other of the keys (ie. speed dial, etc.). Search the code for 30VIP, there are only like 2-3 places where it's referenced. Right. I have done this. It should be immediately obvious how it works. Maybe to some who have been in on the skinny/cisco conversation for awhile. I am not new to c or c++, but am to * and cisco ip phones (this is probably more of my problem) and it is not at all obvious (to me). I would like to possibly contribute, however. I have also been working with chan_sccp which I understand supports these phones more fully than chan_skinny. I am surprised with a $1000+ bounty on the porting of chan_sccp features to chan_skinny that no one has taken time to do it yet. Please forgive me if this sounds ungrateful. I am thankful for the help and for the great product that * is. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speed Dial Application in *
Hi all, Is there a speed dial type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar feature? Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
On 3/30/07, Jason Parker [EMAIL PROTECTED] wrote: - Chris Nighswonger [EMAIL PROTECTED] wrote: On 3/29/07, Jason Parker [EMAIL PROTECTED] wrote: I really don't know what to say then.. It's a simple switch statement on the phone model, with some assignments to set what the buttons do. That much *is* obvious. :P What was not obvious, but I now understand, is how how the loops, etc in the 30VIP case statement related to the actual physical button layout on the phone itself (see one of my previous posts in this thread). After comparing the code of chan_skinny and chan_sccp I was finally able to figure out what was going on there. The chan_sccp code builds a more complete button template for the 30VIP than the chan_skinny. However, there seem to be quite a few other problems with the chan_sccp driver and * 1.4. So, after a few (well commented) hacks I am recompling a slightly modified chan_skinny. We'll see if it works better. Another question about chan_skinny: What code parses the skinny.conf file? Specifically I am looking to set speed-dial softkeys on the 30VIP via skinny.conf. I have not been able to determine if this is already in the code or if I need to keep hacking. Thanks for the help. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
On 3/28/07, Jason Parker [EMAIL PROTECTED] wrote: - Derek Whitten [EMAIL PROTECTED] wrote: if i remember right, most of the buttons on those and the 12SP+ phones don't really work because there isn't a button template in * There is a button template, the problem is that most of the softkeys simply aren't implemented. That is the conclusion I came to and was confirmed today in a very brief chat with one of the individuals listed as a developer on the chan_skinny module. He said that they could be implemented. What I would like to know, and do not understand, is the relationship between the code in chan_skinny.c which sets up the softkeys which are implimented and the actual key positions on the phone. With this info, I can hack the code to impliment other of the keys (ie. speed dial, etc.). Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
Is anyone else on the list using Cisco 30VIP phones with the chan_skinny driver? I have tried to catch the one of the developers on the chat relay, but cannot seem to get anywhere. I am trying to understand how the soft buttons are setup. They are apparently hard-coded into the chan_skinny.c module. Specifically, I am looking for how the code relates to the actual layout of the buttons on the phone. So far, I cannot even get the buttons that are in the code by default to work properly. I have several of these phones up and registered with *. The dialpads work fine. But other buttons do not. Thanks Chris On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote: On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote: I have three registering with * and having basic functionality. I am at a loss to know how to program the buttons (other than dtmf, hold, mute, spkr). Here is what the * console shows when one of the phones registers: -- Starting Skinny session from 192.168.0.70 -- Device 'SEP000196C00CDC' successfully registered Device capability set to '12' Adding button: 9, 1 Adding button: 1, 0 Adding button: 15, 0 Adding button: 126, 0 Adding button: 5, 0 Adding button: 125, 0 It appears that * is setting up some buttons. But where it is getting the config info, I don't know. Sorry for answering my own post, however it may help someone else: Soft button configuration is set in skinny.c I'm still looking for some explaination of the logic and sytax of setting them. Chris -- Chris Nighswonger Network Systems Director Foundations Bible College Seminary www.foundations.edu www.fbcradio.org [EMAIL PROTECTED] V:910-892-8761 C:919-820-5473 - NOTICE: The information contained in this electronic mail message is intended only for the use of the intended recipient, and may also be protected by the Electronic Communications Privacy Act, 18 USC Sections 2510-2521. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please reply to the sender, and delete the original message. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can be called on FreeWorldDialup/IAX channel, but can't make calls
On 3/24/07, Timothy Parez [EMAIL PROTECTED] wrote: Hi, I have an FWD account and it's configured in asterisk. I can be called by people using FWD, but I cannot make FWD calls myself. I have FWD and IAXTel configured as well. FWD has been having problems with their IAX server for awhile judging by their forums. I have exchanged emails with Juan there and after one reboot of their IAX box I was able to call time, echo test, and the likes, but no 800 numbers. I even could receive calls. Then it quit again. Juan reset my account, but it still does not work. IAXTel registers fine. However, I get the conjestion messges with every 800 number I attempt to dial always. But, these are donated services so we can't complain too much. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can be called on FreeWorldDialup/IAX channel, but can't make calls
On 3/24/07, Timothy Parez [EMAIL PROTECTED] wrote: Mar 24 15:28:55 ERROR[2873]: chan_sip.c:11076 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 172.17.249.253, but there is no hint for that extension I believe the subscribe error comes from not having a 'hint' in the context of the extension for the sip @ 172.17.249.253 indicating the sip at extension 00032498043823 (what an extension!). I am new myself to * so someone may need to correct me on this one. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
On 3/22/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 3/22/07, Chris Nighswonger [EMAIL PROTECTED] wrote: 1.4.1 I've got one of those at home and a test system running 1.4.2. I'll take a look tonight and see if there is anything obvious. I'm not a developer, though. I know one of the guys working on chan_skinny uses 30VIPs, so I would have thought it worked. I do know when I tried the 30VIP on chan_sccp, it was doing some weird things. Wondering if you had a chance to look at your 30VIP? I have three registering with * and having basic functionality. I am at a loss to know how to program the buttons (other than dtmf, hold, mute, spkr). Here is what the * console shows when one of the phones registers: -- Starting Skinny session from 192.168.0.70 -- Device 'SEP000196C00CDC' successfully registered Device capability set to '12' Adding button: 9, 1 Adding button: 1, 0 Adding button: 15, 0 Adding button: 126, 0 Adding button: 5, 0 Adding button: 125, 0 It appears that * is setting up some buttons. But where it is getting the config info, I don't know. Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote: I have three registering with * and having basic functionality. I am at a loss to know how to program the buttons (other than dtmf, hold, mute, spkr). Here is what the * console shows when one of the phones registers: -- Starting Skinny session from 192.168.0.70 -- Device 'SEP000196C00CDC' successfully registered Device capability set to '12' Adding button: 9, 1 Adding button: 1, 0 Adding button: 15, 0 Adding button: 126, 0 Adding button: 5, 0 Adding button: 125, 0 It appears that * is setting up some buttons. But where it is getting the config info, I don't know. Sorry for answering my own post, however it may help someone else: Soft button configuration is set in skinny.c I'm still looking for some explaination of the logic and sytax of setting them. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
On 3/22/07, Bill Hackensack [EMAIL PROTECTED] wrote: On 3/21/07, Chris Nighswonger [EMAIL PROTECTED] wrote: I have just successfully configured a Cisco 30VIP to work with my Asterisk server. I have seven of these phones new and would like to deploy them. I am wondering if anyone has this phone deployed with Asterisk and can suggest configuration of the various buttons, etc. (Bare with me as I am new to Asterisk.) So which is it? You either have it configured or you don't. Sorry. Configured is a relative term in my previous post. From the standpoint of the phone making calls it is configured. From the standpoint of buttons other than the dialpad working (ie 'xfer' 'dspl' 'hold' 'redl' the 26 soft keys), it is not configured. For example, pressing the 'hold' key in the current configuration causes Asterisk to crash very bad I suspect. Also, the display shows time and occasionally an 's' Hope this clarifies my request. Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
On 3/22/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 3/21/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Hi all, I have just successfully configured a Cisco 30VIP to work with my Asterisk server. I have seven of these phones new and would like to deploy them. I am wondering if anyone has this phone deployed with Asterisk and can suggest configuration of the various buttons, etc. Which version of Asterisk? 1.4.1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 30VIP Phone
On 3/22/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 3/22/07, Chris Nighswonger [EMAIL PROTECTED] wrote: 1.4.1 I've got one of those at home and a test system running 1.4.2. I'll take a look tonight and see if there is anything obvious. I'm not a developer, though. I know one of the guys working on chan_skinny uses 30VIPs, so I would have thought it worked. I do know when I tried the 30VIP on chan_sccp, it was doing some weird things. Here is some clarification from testing today 1. Pressing the 'Hold' button with the phone on-hook cause * to crash with a Segmentation Fault. 2. Pressing the 'Hold' button with the phone off-hook generates the following on the * console: [Mar 22 13:23:53] WARNING[7200]: channel.c:2048 ast_waitfordigit_full: Unexpected control subclass '16' [Mar 22 13:23:54] WARNING[7200]: channel.c:2048 ast_waitfordigit_full: Unexpected control subclass '17' [Mar 22 13:23:54] ERROR[7193]: chan_skinny.c:3638 handle_open_receive_channel_ack_message: No RTP structure, this is very bad 3. Pressing the 'Hold' button with the phone off-hook and in a call results in expected behavior. Perhaps there is something incorrect in my skinny.conf so here is a snip: [chris] device=SEP000196C00D3F version=P002L2J2; Thanks critch context=local line = 51235 ; Dial(Skinny/[EMAIL PROTECTED]) Thanks for the help. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 30VIP Phone
Hi all, I have just successfully configured a Cisco 30VIP to work with my Asterisk server. I have seven of these phones new and would like to deploy them. I am wondering if anyone has this phone deployed with Asterisk and can suggest configuration of the various buttons, etc. (Bare with me as I am new to Asterisk.) Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error compiling zaptel 1.4.0
Hi all, I decided the best way to get to know * well is to do it from scratch. Having read the majority of the Asterisk: The Future of Telephony I am now attempting to compile zaptel 1.4.0 and am receving the very same series of errors mentioned in this post on the forums: http://forums.digium.com/viewtopic.php?t=13619highlight=zaptel1+++zttranscode++error However, there has been no solution offered to this issue. I am running 2.6.20-1.2925.fc6. I have combed through the various suggestions for issues compiling zaptel. I do have the kernel-devel package installed and the proper sym links, etc. When performing 'make linux26' it blows chunks trying to compile zttranscode.o. Here is what it looks like: CC [M] /usr/src/zaptel-1.4.0/zttranscode.o /usr/src/zaptel-1.4.0/zttranscode.c: In function 'zt_tc_open': /usr/src/zaptel-1.4.0/zttranscode.c:192: error: invalid use of undefined type 'struct page' /usr/src/zaptel-1.4.0/zttranscode.c:193: error: invalid use of undefined type 'struct page' /usr/src/zaptel-1.4.0/zttranscode.c:194: error: increment of pointer to unknown structure /usr/src/zaptel-1.4.0/zttranscode.c:194: error: arithmetic on pointer to an incomplete type /usr/src/zaptel-1.4.0/zttranscode.c:195: error: implicit declaration of function 'SetPageReserved' /usr/src/zaptel-1.4.0/zttranscode.c: In function 'ztc_release': /usr/src/zaptel-1.4.0/zttranscode.c:208: error: invalid use of undefined type 'struct page' /usr/src/zaptel-1.4.0/zttranscode.c:209: error: invalid use of undefined type 'struct page' /usr/src/zaptel-1.4.0/zttranscode.c:210: error: increment of pointer to unknown structure /usr/src/zaptel-1.4.0/zttranscode.c:210: error: arithmetic on pointer to an incomplete type /usr/src/zaptel-1.4.0/zttranscode.c:211: error: implicit declaration of function 'ClearPageReserved' /usr/src/zaptel-1.4.0/zttranscode.c: In function 'zt_tc_mmap': /usr/src/zaptel-1.4.0/zttranscode.c:370: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.0/zttranscode.c:376: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.0/zttranscode.c:376: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.0/zttranscode.c:378: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.0/zttranscode.c:378: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.0/zttranscode.c:384: error: implicit declaration of function 'remap_pfn_range' /usr/src/zaptel-1.4.0/zttranscode.c:384: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.0/zttranscode.c:384: error: 'PAGE_SHARED' undeclared (first use in this function) /usr/src/zaptel-1.4.0/zttranscode.c:384: error: (Each undeclared identifier is reported only once /usr/src/zaptel-1.4.0/zttranscode.c:384: error: for each function it appears in.) make[2]: *** [/usr/src/zaptel-1.4.0/zttranscode.o] Error 1 make[1]: *** [_module_/usr/src/zaptel-1.4.0] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.20-1.2925.fc6-i686' make: *** [linux26] Error 2 and dumps me back to the prompt. I am working with a fresh install of fc6. Any help is appriciated. Chris -- Chris Nighswonger Network Systems Director Foundations Bible College Seminary www.foundations.edu www.fbcradio.org [EMAIL PROTECTED] V:910-892-8761 C:919-820-5473 - NOTICE: The information contained in this electronic mail message is intended only for the use of the intended recipient, and may also be protected by the Electronic Communications Privacy Act, 18 USC Sections 2510-2521. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please reply to the sender, and delete the original message. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling zaptel 1.4.0
On 3/16/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: Chris Nighswonger wrote: I am working with a fresh install of fc6. Kernel 2.6.20 was released after Zaptel 1.4.0, so it will not build against that kernel. Either use an older kernel, use the SVN version of Zaptel branch-1.4, or wait for the release of Zaptel 1.4.1 which should occur later today. I used the SVN version and it compiled without complaint! Thanks Kevin. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
On 3/8/07, Chris Nighswonger [EMAIL PROTECTED] wrote: Thanks for the responses. iptables on the * box has no rules and all tables default to 'accept.' I have not got to the point of placing calls out across the internet yet. The issue here is no audio back from the * box when running through the demo routine. I'll try to set it up to make a call outside tomorrow. Ok. I have not been able to setup the box to call outside, however, watching the packet traffic I see plenty of data flowing from the xlite client to the * server, but never any packets from the server to the client. (That is, during the course of the call.) The server and client talk just fine when establishing the connection, just no audio data from the server to the client. Any thoughts? From everything I've read, the initial setup should be much easier than mine has gone so far... :( Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie Question
Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question
Thanks for the responses. iptables on the * box has no rules and all tables default to 'accept.' I have not got to the point of placing calls out across the internet yet. The issue here is no audio back from the * box when running through the demo routine. I'll try to set it up to make a call outside tomorrow. Chris On 3/8/07, Leonardo Kamache (Gmail) [EMAIL PROTECTED] wrote: Don't forget about 4569 UDP port (IAX protocol) forwarded to your Asterisk box. Best Regards; Leonardo Kamache On 3/8/07, Dovid B [EMAIL PROTECTED] wrote: If both the asterisk server and the softphone are on the same LAN then I would look at your firewall settings on the box. Make sure you have 5060 and 10,000 - 20,000 UDP open. If the phone is connecting to the server over the internet and the server IS behind NAT then you need to forward ports 5060 and 10,000-20,000 UDP to the asterisk server. - Original Message - From: Chris Nighswonger [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 09, 2007 1:16 AM Subject: [asterisk-users] Newbie Question Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid=Test Unit 1234 host=dynamic; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw [EMAIL PROTECTED]; Subscribe to status of multiple mailboxes context=internal Here is the problem: Xlite registers fine. When I dial 500 to access the demo, the * console shows the client connect and the demo audio plays. However, there is no sound on the client end. I have installed Xlite on an XP workstation and on a *nix workstation. Both installs behave the same. Any thoughts? Or do I need to post more details? Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chris Nighswonger Network Systems Director Foundations Bible College Seminary www.foundations.edu www.fbcradio.org [EMAIL PROTECTED] V:910-892-8761 C:919-820-5473 - NOTICE: The information contained in this electronic mail message is intended only for the use of the intended recipient, and may also be protected by the Electronic Communications Privacy Act, 18 USC Sections 2510-2521. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please reply to the sender, and delete the original message. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
