Re: [asterisk-users] Recommendations for IMAP Voicemail
On 05/05/15 17:52, Olivier wrote: 2. From personal experience, would you rate an IMAP migration as an easy or as a difficult task ? By IMAP migration, I mean changing from one IMAP software to another, on the same or on an other box. There is software called 'imapsync' which will sync mail from one imap store over to another. I have used that to migrate from one server to another and it was pretty painless. It took a while to do but it does it a mail at a time so it is something you can run while everyone is using the system and then right at the end you just need to do a final sync while voicemail is disabled. -- _ -- 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] Recommendations for IMAP Voicemail
With regard to your question about the imap store I highly suspect that if it is unavailable when the message is saved then the message will be lost. There are configuration options for the timeout and it might retry a couple of times but Asterisk has no scheduling so I cannot see a way for it to record the fact that there are voicemails waiting to be uploaded. You are concerned about reliability but this is not an easy thing to fully plan for. You could have a couple of asterisk boxes with config stored in a database which is replicated between two of them. For voicemail and general audio storage this could be on a NAS which replicates data over to a backup. However if the primary NAS fails then because you are using NFS this has a tendancy to hang which will cause problems. You could use IMAP for storage but you are pushing the file storage issues just furthur away. If you use an IMAP store based on MAILDIR++ then you can have multiple IMAP servers mounting the same data and it will work fine however this approach typically has a poor performance when it comes to stat files in a mailbox so if you have lots of voicemails asterisk will pause a long time. Other systems such as Cyrus which use a database are much faster (sometimes 100 times faster!) but much slower when it comes to storing new files. We have a redundant pair of opensips servers performing load balancing between multiple asterisk boxes which have a large gluster file storage mounted on each. Thats a large number of machines but is the sort of topology you need for something to be fully fault tolerant. On 06/05/15 15:15, Olivier wrote: imapsync seems a very interesting tool. Thanks for sharing this. Now I'm still curious to learn if moving from local file storage to cloud IMAP storage is still resilient to short network outages (without IMAP replication). Regards 2015-05-06 10:18 GMT+02:00 Gareth Blades mailinglist+aster...@dns99.co.uk mailto:mailinglist+aster...@dns99.co.uk: On 05/05/15 17:52, Olivier wrote: 2. From personal experience, would you rate an IMAP migration as an easy or as a difficult task ? By IMAP migration, I mean changing from one IMAP software to another, on the same or on an other box. There is software called 'imapsync' which will sync mail from one imap store over to another. I have used that to migrate from one server to another and it was pretty painless. It took a while to do but it does it a mail at a time so it is something you can run while everyone is using the system and then right at the end you just need to do a final sync while voicemail is disabled. -- _ -- 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 between snom 300 and aastra 6731i
You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: please no body has som with aastra can help me in this issue 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com mailto:salah.elharit...@gmail.com: hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ok inbound and outbound the calls between x-lite and snom300 ok inbound and outbound the issue just between snom and aastra i can call from aastra to snom without issue but when itry to call from snom300 to aastra6731i i get bad request all the time i test with 3 snom300 i get the same result please any body have the snom and aastra can help me in order to fixe this issue thanks and 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] call between snom 300 and aastra 6731i
That is your issue. You can enable a 'sip debug' and make the call again and get a trace of the SIP message asterisk is sending to the phone. We can take a look here to see if anything looks wrong. If you could post a trace from a phone that can call that destination it might be easier to spot the differences. I am off work for the weekend now but someone else might be able to take a look in the next couple of days. On 27/03/15 17:44, Salaheddine Elharit wrote: -- Called SIP/FD/00XX17621 -- Got SIP response 480 No address found back from 217.195.XX.XXX:5060 -- _ -- 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] 1.8.11.0 - CLI error res_timing_timerfd
On 12/02/15 12:25, Stefan Viljoen wrote: Hi all Sometimes (about every three months) some of my Asterisk 1.8 boxes will start running this message thousands of times in the CLI: [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument ad infinitum Sometimes, this will disappear after a few minutes, and not re-appear. On other occassions, after this started appearing in the CLI, the Asterisk will crash without any slowdown or problems. On yet other occassions, the box will start getting slower, load average rising and rising until it become virtually frozen and it has to be rebooted in order to come back up in a functional state again. I've also noticed that stopping the Asterisk process and restarting it does not help if this error is in effect and a slowdown (rising load average) has started - the whole physical machine has to be rebooted in toto. I'm running on Centos 6.5. Anybody else seen this message before? What does it mean? Thanks Kind regards Stefan timerfd is the kernel based timing source. If restarting Asterisk does not fix it then the next step I would suggest is updating the system to make sure you have the latest kernel installed and giving the server a reboot to activate it. We run Asterisk 11.6 on Centos 6 and have never seen that issue on any of our servers. -- _ -- 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] Question regarding custom announcements used by several Asterisk servers
On 06/02/15 07:54, Olli Heiskanen wrote: My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? I assume you mean NFS. Yes you can do that although using NFS you will then have a single point of failure and in the standard NFS client configuration if you try to access a file which is on NFS but it is unavailable then the file access will hang. So you might be better off having the files copied onto each of the asterisks servers local file storage or use a redundant file system such as gluster. -- _ -- 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] constantly increasing load in Asterisk 11.14
On 05/02/15 10:53, Sebastian Damm wrote: Hi, we have quite a few Asterisk machines running and try to keep them on a current version of the Asterisk 11 branch. But since we upgraded to 11.14.0 a couple weeks ago, we have to restart the Asterisk process every week because the load gets too high and our monitoring complains. Those machines are doing only SIP-to-SIP call relay, the dialplan is quite complex, transcoding is done only on a few percent of the calls processed. During the daytime, there are at max around 200 SIP channels (100 calls) running at the same time. After one week, one machine has processed about 170k calls. I have uploaded a comparison of cacti load graphs for one week of a machine running with 11.14.0 and one running with 11.6.0: http://pbrd.co/1v0SO3R As you can see, after a restart, both machines have about the same load. But after the really quiet weekend, the 11.14 Asterisk starts the new week with a much higer load than the 11.6 Asterisk, where it stays constant. We've had an 11.5.1 machine running for about half a year without the need of restarting, but right now, this is not possible. Has anyone seen this before? Or does anyone know a reason, what change somewhere between 11.6 and 11.14 could cause this behaviour? It looks like we have to go back to 11.6. We see it in the certified version which is currently 11.6. We have about 50% higher call volume on our servers than yourselves and it does take about 3-4 months for the load to reach about 5. I don't know what causes it but we take the servers out of our load balancing and restart asterisk every few months and this returns the load to normal. -- _ -- 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] Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
On 05/12/14 16:46, Olli Heiskanen wrote: INVITE that Asterisk (at port 5070) receives: PU.BL.IC.IP:5060 PU.BL.IC.IP:5070: SIP, length: 1046 INVITE sip:6...@testers.com mailto:sip%3a...@testers.com;transport=UDP SIP/2.0 Record-Route: sip:PU.BL.IC.IP;lr=on;ftag=41030177 Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0 Via: SIP/2.0/UDP AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z- Max-Forwards: 69 Contact: sip:7...@ast.er.isk.ip:38699;transport=UDP To: sip:6...@testers.com mailto:sip%3a...@testers.com;transport=UDP From: 771sip:7...@testers.com mailto:sip%3a...@testers.com;transport=UDP;tag=41030177 Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.2.21357 r21367 Allow-Events: presence, kpml Content-Length: 239 v=0 o=Z 0 0 IN IP4 AST.ER.ISK.IP s=Z c=IN IP4 AST.ER.ISK.IP t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv This client is saying it only supports speex and iLBC and would prefer them in that order. Your sip.conf appears to only permit alaw, ulaw and gsm so there is no mutual supported codec and hence the call fails. -- _ -- 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] Hangup Chanel when a peer unregisters
On 04/11/14 15:11, Pat Collins wrote: Hello group and thank you for the attention. I'm using Asterisk 11.12 running on Ubuntu Server 12.04 We have an issue with channels remaining open after a SIP peer unregisters. It seems that if the peer goes away before manually hanging up a call, the channel remains open until a hangup request is sent from the CLI. Is there any way to drop a channel when the peer using it disappears? Googled every phrase I could think of. No luck. Thank you! Pat Collins rtptimeout= in sip.conf will hangup a channel if no rtp is received for a period of time. -- _ -- 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] PlayTones not working
On 30/10/14 19:40, Henry Fernandes wrote: I’m trying to use Playtones to have a tone played periodically throughout phone calls. Unfortunately, I can’t seem to get PlayTones to work. I never hear the audio tones. Here is the output on the Asterisk console. -- Executing [19525553312@proxy-dial:2] *PlayTones*(*SIP/testphone-0032*, *1400/500,2000/5000*) in new stack [2014-10-30 14:28:31] *WARNING*[23154]: *translate.c*:*206* *framein*: no samples for ulawtolin -- Executing [1952553312@proxy-dial:3] *Dial*(*SIP/testphone-0032*, *SIP/19525553312@proxy01,,gU(record_call_id)*) in new stack I’ve checked the debug log and I can’t see any related errors or warning beyond the one above. -H Has the call already been answered? If not make sure you call Progress() to enable early audio. -- _ -- 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] PlayTones while in call
In that case the only way I can think of doing it would be to place both parties into a conference call and have an extension join it which just plays a tone into the conference every so often. On 31/10/14 16:40, Henry Fernandes wrote: Unfortunately, the majority of my customers are using 1.8. We do have some customers on Asterisk 11, but that doesn’t have this option either. I looked at the changes for this feature (https://gitorious.org/asterisk-tools/asterisk/commit/14cd18d04125ebd8b78e9aefcb2c98eb48b741d4). Unfortunately, it doesn’t look like it will be straightforward to back port it to either Asterisk 11 or 1.8. -H From: John Kiniston johnkinis...@gmail.com mailto:johnkinis...@gmail.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Date: Friday, October 31, 2014 at 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PlayTones while in call Henry, Both Montior and MixMonitor have a 'B' option that plays a periodic tone. B([interval]): Play a periodic beep while this call is being recorded. interval - Interval, in seconds. Default is 15. https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Monitor https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MixMonitor On Fri, Oct 31, 2014 at 8:26 AM, Henry Fernandes he...@usinternet.com mailto:he...@usinternet.com wrote: I’ve gotten PlayTones to work, however it stops playing the tones as soon as the call is answered. I would like to use PlayTones during the call because I want to have a tone/beep played in the background while call recording is going on. Anyone know a way to get PlayTones to work while call is in progress? Alternatively, does anyone have a suggestion for playing the tone/beep for recorded calls. -H -- _ -- 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 -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] ${HASH(SIP_CAUSE,channel-name)}
On 30/10/14 13:52, Jonas Kellens wrote:
Hello,
I read on the wiki :
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
*${HASH(SIP_CAUSE,channel-name)}*. Additionally make sure you're
using the destination channel, not the source channel.
But when I use this in my dialplan, this 'variable' is empty.
Dialplan :
exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,CHANNEL)})
CLI :
[Oct 30 14:48:03] -- Executing [h@pbx-routing:5]
NoOp(SIP/SipAT01-0015, sip cause = ) in new stack
[Oct 30 14:48:03] -- Executing [h@pbx-routing:6]
NoOp(SIP/SipAT01-0015, sip cause = ) in new stack
Take a look at my blog entry about it :-
http://gblades.blogspot.co.uk/2013/07/how-to-get-sip-response-code-in.html
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Re: [asterisk-users] Asterisk support for Bittorrent Bleep
On 11/08/14 16:46, Farid Fadaie wrote: Hello, Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent Bleep (a private P2P SIP-based messaging application in early alpha) http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-communications/ I have personally been a fan of Asterisk and have been using it for years and now that we have (kind of) released Bleep, I wanted to ask you guys to let us know what you think. Considering that Bleep is built on an engine (think of it as a distributed SIP proxy) that supports SIP, I thought it might be beneficial to ask you guys for your ideas. Here is what I have in mind but will be happy to hear your thoughts on everything that is relevant to Bleep and Asterisk: 1- What do you think about supporting Bleep in Asterisk? Similar to Skype channels but way more flexible (considering the interface will be SIP). Our engine can take care of all lookups, NAT traversals, encryption, etc. We can essentially enable Asterisk connected devices to be able to talk to Bleep users. 2- How could the Asterisk community benefit from Bleep (or the engine behind it)? 3- what features would you like to see implemented in Bleep (the consumer app) or its engine? Let's see if we can come up with any interesting idea. Thanks in advance. Farid Given how many SIP based attacks and probes are going around and that most peoples defence is to use something like fail2ban I think your biggest hurdle in getting people to use it will be persuading them that it is secure and giving them the tools to block malicious use. -- _ -- 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] Terrible dahdi_test results
On 15/05/14 16:28, Mike Leddy wrote: Hi Russ, I rebooted the machine loading dahdi_dummy in /etc/modules before the /etc/init.d/dahdi. Now dahdi_test shows a nearly perfect score: # dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.990% 99.998% 99.996% 99.998% 99.998% 99.997% 99.997% 99.998% 99.997% 99.998% 99.997% 99.998% 99.998% 99.997% 99.998% 99.997% 99.997% 99.997% 99.998% 99.997% 99.998% 99.998% 99.997% ^C --- Results after 24 passes --- Best: 99.998% -- Worst: 99.990% -- Average: 99.997188% Cummulative Accuracy (not per pass): 99.997 When I connect a live E1 to the card it does work but I get a fair number of: [May 15 15:10:39] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:10:42] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:10:45] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:11:01] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:11:01] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:11:06] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:11:12] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:11:12] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:11:12] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 [May 15 15:11:13] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 Resulting in: [May 15 15:10:34] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 [May 15 15:10:39] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 [May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 [May 15 15:10:44] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 [May 15 15:10:45] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 [May 15 15:11:13] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 [May 15 15:11:17] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 [May 15 15:11:22] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on D-channel of span 1 Not usable in production but getting a lot closer. Is there anything else that can be done to improve this ? Best regards, Mike Check your span= line in you configuration. If your telco is providing clocking and you are set to generate it yourself then they go out of sync which generally causes errors like these. If you are set to be the clock master try changing it to see if it improves. It should either improve or not work at all. -- _ -- 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] Ringing issue
You would need to provide more information. Mobiles and landlines are not SIP and yet you say calls are coming into your asterisk over SIP. So what or who is doing the translation? Initial thoughts are that it could be you are sending back SIP/180 with no session progress and indicating ringing but the other end is misconfiguration and not generating its own ring tone. This is possible if you have multiple providers sending you calls or one provider using different kit for different geographic areas. On 13/05/14 12:01, D'Arcy J.M. Cain wrote: I have an issue with ringing. Some users who call my switch hear ringing and others don't. I have researched this and understand the issue of firewalling and RTP. My switch has UDP ports 1 to 2 open. In any case I think that blocked RTP would block all ringing, not just some. I have one origination provider. As far as I can tell the issue is related to the remote user's provider. My sister does not hear ringing when she calls from her Roger's cell phone but she does from her Vonage phone. I hear ringing when calling in from my Koodo cell phone. Some land lines work and others do not. The server is not behind a NAT and neither is the origination provider. There is a firewall but port 5060 is open (UDP and, just in case, TCP) as well as the RTP ports mentioned above. I am not sure where to look next. I assume that there is some sort of signaling that I am not doing but I can't figure out where. Can anyone suggest what area I should be looking? Thanks. -- _ -- 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] High Availability with Asterisk
On 07/03/14 16:52, Johann Steinwendtner wrote: Sorry, for the stupid question, but what happens if Kamailio fails ? We have two copies on different servers which make use of keepalived to provide a virtual IP address between them. We also have them connected to two databases with active-active replication. -- _ -- 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] Hacking attempt, Asterisk 1.4
On 20/02/14 11:27, Brynjolfur Thorvardsson wrote:
Hi all
We have an Asterisk server that's been running for a few years now
without problems. We have IPTables running, as well as fail2ban and
have followed all the security recommendations we have found.
Every few weeks we get an attack that lasts about a minute or two,
resulting in our AGI script being overloaded.
What happens is that somebody seems to be trying to connect from our
server -- in my cdrs log I can see that they use a four digit number
for source, destination and caller id, e.g.
clid: 7321
src: 7321
dst: 7321
channel: SIP/xx.xx.xx.xx-
xx.xx.xx.xx is our server IP. When one of our registered users makes a
call the channel is SIP/- where is the SIP
user ID.
So it looks like a SIP phone trying to call itself, using our Asterisk
server IP as SIP user name.
Within a couple of minutes the attacker seems to go through some 1
attempts, resulting in our AGI script collapsing from the load. My
Asterisk full log shows something like:
-- Executing [7321@sip:1] Answer(SIP/xx.xx.xx.xx-b0828f20, )
in new stack
-- Executing [7321@sip:2] AGI(SIP/ xx.xx.xx.xx -b0828f20,
agi:// xx.xx.xx.xx ) in new stack
-- Executing [7321@sip:3] Hangup(SIP/ xx.xx.xx.xx -b6130f70, )
in new stack
== Spawn extension (sip, 7321, 3) exited non-zero on 'SIP/
xx.xx.xx.xx -b6130f70'
cdr_odbc: Query Successful!
-- AGI Script agi:// xx.xx.xx.xx completed, returning 0
Our AGI script refuses to call illegal numbers, while our Asterisk
dialplan is a bit more accommodating, mostly because I have had
problems figuring out the order in which to put the various rules (I
might have another look at that!)
Does anybody know how to stop this from happening -- I can't find the
attackers IP number in my logs, and these attacks happen infrequently,
and are over quickly, so that I haven't had an opportunity to run sip
debug during an attack, and I don't want to have it running all the time.
Best regards
Binni
Brynjólfur Þorvarðsson
IT Consultant
Tlf. +45 88321688
I have this in my extensions.conf :-
[default]
; all unauthenticated connection attempts from the internet come in here.
exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
${SIP_HEADER(Contact)})
exten = _[+*#0-9].,n,Congestion
Then in fail2ban I have the extra line added to the failregex so it is
now :-
failregex = Registration from .* failed for \'HOST\' - Wrong password
Registration from .* failed for \'HOST\' - No matching
Unauthenticated call attempt .*\@HOST\:
That seems to work pretty well for me. Assuming the attacks are
unauthenticated why are you accepting them and running an AGI script and
not rejecting them earlier?
If you need to allow anonymous inbound calls (which is required in some
cases) then I would have the AGI detect them and write an output to
verbose() with the SIP_HEADER(Contact) or any other header which
correctly indicated the origin of the packet.
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Re: [asterisk-users] Variables are empty after Redirecting a channel
On 20/02/14 10:24, Igor Dvorzhak wrote:
Guys,
I am using
Asterisk 1.8.20.0 built by mockbuild @
buildvm-24.phx2.fedoraproject.org
http://buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux
on 2013-01-18 19:52:25 UTC
How can I set variable in one context and then Redirect a channel to
another context and use variable there? The code below doesn't work,
so I've got empty VAR1 in context_2
[context_1]
exten = s,1,SET(__VAR1=VALUE1)
exten = s,n,ChannelRedirect(${CHANNEL},context_2,AMD,1)
[context_2]
exten = AMD,1,NoOp(VAR1: ${VAR1})
Thank you in advance,
Igor
You should be able to get something working using a shared variable. I
have used them in one of my blog posts if you would like an example of
their use
http://gblades.blogspot.co.uk/2013/07/how-to-get-sip-response-code-in.html
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Re: [asterisk-users] G729 - what happens if licences used up?
On 20/02/14 17:16, Paul Belanger wrote: On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifieldt...@softins.co.uk wrote: I haven't been able to find the answer online, and am not currently able to conduct an experiment to find the answer... I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there is a need to perform transcoding, e.g. play a non-g729 sound, or do voicemail, or enter a Meetme, etc. What happens when a SIP call in progress needs a G.729 licence and they are all in use already? Does the call fail, or go silent, or do a re-INVITE to negotiate another codec? I'm interested in what happens on Asterisk 1.2 (for a legacy system), and also whether it is any different on later versions. The question depends if you are offering up other codecs or not. If you only using g729, the call will fail to establish because lack of codecs. If you offer a both g729 and ulaw, then ulaw will be used. That would only apply for new calls. Even new calls would still typically accept g729 even if there are no licenses remaining as there might not be transcoding required. What I would expect to happen if there were no licenses is for you to see an error on the console (possibly repeated multiple times) and for there to be no audio. This is certainly what happens if you have a g729 call with no license and then try to play a sound file which does not have a native g729 format. -- _ -- 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] ConfBridge speak wave file in conf
On 15/02/14 20:05, Jerry Geis wrote: I have a confbridge in asterisk 11. I am using an AGI to bring people in the conf automatically. I want to speak a pre-recorded wave file message into the conf to all users. how might I do that? Thanks, Jerry You could initiate a call which would connect one end to the conference and then the other would pay the message and then hang up. You could establish a call either via the asterisk manager interface or have the agi create a .call file in a temp directory and then move it across to where asterisk looks for 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
Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?
On 14/02/14 06:33, Daniel van den Berg wrote: Hi All, Lets say I want to setup a queue that will handle inbound calls to dynamically added agents that are all mobile numbers. Now when I do this setup it works, it loads the agents dynamically and if the mobile phone is on and have reception it works. But when the phone is for arguments sake off or dont have reception it goes to voice mail for that mobile phone. I don't want this to happen...:) I would like for the queue to continue ringing until there is a time out specified which then takes the caller out of the queue and to voice mail which I then intend to mail somewhere. I guess my question is can this be done in Asterisk? Can I force clients in this queue not to leave a voice message on the mobile phone but rather the Asterisk system? Because when the mobile phone which is an agent in the queue goes to voice mail it answers the call and then plays the voice mail message. My initial thoughts are to maybe ask the mobile operator to switch off the voice mail functionality on those mobile phones and rather give a busy or engaged tone, but I would rather want to do this in Asterisk. Any help or advise on this matter will be greatly appreciated. Thanks! Daniel van den Berg SureTel - South Africa I would suggest using the 'M' option on the Dial command to run a macro. The macro can just wait fir a key to be pressed and until it is pressed the Dial is still effectively ringing. So if it does go to voicemail then the call wont get put through. You need to make sure you have a suitable value set to abandon the agent call if its ringing too long. The callee may also find they are left multiple voicemail messages. -- _ -- 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] g726 transcoding
On 11/02/14 18:45, Dave Platt wrote: Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? Are the modules actually loaded? Try doing a module show and see if the codec modules actually show up as having been loaded. If not check your modules.conf file and see if they've been disabled, and check your Asterisk modules directory to confirm that they were actually installed. Yes that was the problem. I posted an update yesterday but it got sent from the wrong account so didnt make it to the list. When I installed the system I had disabled a lot of the modules from load which were not used and the codecs were part of that. -- _ -- 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] g726 transcoding
Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? I am running certified-asterisk-11.2-cert2 Thanks Gareth core show translation paths alaw --- Translation paths SRC Codec alaw sample rate 8000 --- alaw To g723 : No Translation Path alaw To gsm : (alaw)-(slin)-(gsm) alaw To ulaw : (alaw)-(ulaw) alaw To g726 : No Translation Path alaw To adpcm : No Translation Path alaw To slin : (alaw)-(slin) alaw To lpc10 : No Translation Path alaw To g729 : No Translation Path alaw To speex : (alaw)-(slin)-(speex) alaw To speex16 : (alaw)-(slin)-(slin16)-(speex16) alaw To ilbc : No Translation Path alaw To g726aal2 : No Translation Path alaw To g722 : No Translation Path alaw To slin16: (alaw)-(slin)-(slin16) alaw To siren7: No Translation Path alaw To siren14 : No Translation Path alaw To testlaw : (alaw)-(slin)-(testlaw) alaw To g719 : No Translation Path alaw To speex32 : (alaw)-(slin)-(slin32)-(speex32) alaw To slin12: (alaw)-(slin)-(slin12) alaw To slin24: (alaw)-(slin)-(slin24) alaw To slin32: (alaw)-(slin)-(slin32) alaw To slin44: (alaw)-(slin)-(slin44) alaw To slin48: (alaw)-(slin)-(slin48) alaw To slin96: (alaw)-(slin)-(slin96) alaw To slin192 : (alaw)-(slin)-(slin192) -- _ -- 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] Connect to remote GW
On 04/02/14 18:56, Meadows Hoa wrote: If SIP channel driver needs to connect to a remote GW over a dedicated SIP trunk BUT the remote GW has a 'standby' in case of failure, how can the sip configuration file be configured for the remote GW when there are actually two IP addresses. If the main remote GW fails control automatically switches to the standby GW, so how could the SIP configuration file hande this switch and support both host IP addresses. There is no DNS so straight IP addressing is used. Doing it in the dialplan is a bit of a bodge but perfectly possible. Just make sure you have qualify=yes so if a GW does down asterisk spots it and the Dial() command returns quickly. You might also wish to reduce the check interval down from the default of every minute. The proper way is to make use of DVS SRV however Asterisk doesnt support it properly (doesnt fail over). Not sure if that has been fixed in version 12. If there is just one active GW and one or more backups then the its normally fairly easy to create a virtual IP address on the GW so if one fails then another takes over. Only works if they are on the same network though. Alternatively use something like opensips as a front end. Thats what we do and it load balances between multiple asterisk server and detects if one fails. You can make use of keepalived to provide a virtual IP address which moves between boxes if one dies or the opensips process stops running. -- _ -- 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] Repeated Locally bridging messages
We have a customer reporting poor quality calls when they come to us via a particular provider. The SIP traces look perfectly normal both on the ingress to us and egress to another telco. No additional sip messages after the call has been answered until the BYE is received. However in the asterisk logs I am getting this :- 2014-02-05 13:45:03 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:05 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:05 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:05 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:06 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 Any idea what could be causing this? I am running asterisk 11.2-cert2. I am going to get call redirected via our test box and turn on full verbosity in the logs and capture a full tcpdump but any ideas would be welcome. Thanks Gareth -- _ -- 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] Telco with multipe SIP servers
On 02/02/14 14:42, Markus Reschke wrote: Hi! My telco is Deutsche Telekom and they got about 30 SIP servers right now. Currently I've set up a template for incoming calls in sip.conf and added each SIP server by it's IP address like this: [DTAG-in-1](DTAG-in-template) host=217.0.16.103 ... [DTAG-in-30](DTAG-in-template) host=217.0.20.99 I've done that to improve security and to be able to assign all calls coming in via Deutsche Telekom to a dedicated dialplan context. Unfortunately this approach is not scalable and it's a PITA to maintain a list of server IP addresses since Deutsche Telekom will get more SIP servers in the future. They've started to migrate the classic POTS/ISDN network to VoIP, the goal is get it done by 2016. Customers with DSL get VoIP directly, i.e. they need SIP phones or a SIP PBX, and customers with a phone line only are converted by the MSAN. And they don't provide an official list of the SIP servers :-( By some reverse engineering I found out that all SIP servers are within a specific subnet. Is there any way to match peers by subnet(s) instead of FQDNs or single IP addresses? If not, it would be a feature really needed to be able to cope with telcos running multiple or tons of SIP servers. cu, Markus You could consider making use of opensips. We use it for inbound sip connections and its fairly easy to get it to perform a database lookup against a connecting IP address and pull out a record and pass that onto Asterisk using a custom header. Asterisk can then trust connections from opensips and you can read in the custom header and have the dialplan decide what to do based upon the value. -- _ -- 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] dimensioning
On 28/01/14 15:01, Jerry Geis wrote: I have been trying to get a feel for scaling or dimensioning using asterisk 11. if I desire to use something like a dell r320, hardware RAID, 2G E5-2420, 4G RAM and only SIP trunking using gsm (least bandwidth and no transcoding) how many calls out can I expect to make at one time and asterisk still be OK and responsive? Thanks, Jerry We have a quad 'Intel(R) Xeon(R) CPU E5-1410 0 @ 2.80GHz' that will copy with about 800 concurrent calls with a lot of AGI stuff the log call start and end and about 5-10% of the calls being recorded. All using g711alaw. All calls come in and go out via sip. -- _ -- 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] [HELP]: Auto-answering calls placed from call files
On 28/01/14 16:56, Steve McCann wrote: Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm trying to have an automated call to an Aastra SIP phone and have the call auto-answeredby the phone. I know that a SIP call placed to the phone can be auto-answered if a certain SIP header is added to the call. I am able to apply the SIP headers manually and get that working (using SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but for call files, I don’t seem to be able to edit any of the sip headers - there is only basic customizations allowed to setup the calls. Does anyone know how I could place automated outgoing calls that would have the proper sip headers added to it that would allow the call to be auto-answered? I've also posted this question to the forums here: http://forums.asterisk.org/viewtopic.php?f=1t=89190 http://forums.asterisk.org/viewtopic.php?f=1t=89190 Many thanks, Steve So I take it in the call file you have it set to call Dial(SIP/something) ? If rather than dialling the sip destination immediately you dialled a local channel then it could add the custom header and then initiate the dial to the sip destination. -- _ -- 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] mixmonitor extension
On 23/01/14 23:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? what about patch for Opus? uncle google doesnt know MP3 is only supported for reading not writing. Its a patent issue as Asterisk cannot distribute the software to write to mp3 under its own license. Its a similar issue with Opus as the codec is covered by a couple of patents in the USA. What most people do is use MixMonitor to record to .wav (alaw) and then in the 'h' extension call a program which runs a background task to convert the .wav file to whatever format they wish. Thats what we do but we actually use the Monitor application and we end up with both legs of the call and multiple sets of recordings if people pause and unpause. We then move these files off to a different server when they get mixed and converted to mp3 and then emailed out to our customers. We do it this way to reduce the load on the Asterisk boxes but also keep all the call recordings in a central location. -- _ -- 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] Change the preferred audio playback format
On 23/01/14 13:38, Ishfaq Malik wrote: Hi Is there any way to change the preferred audio playback format in asterisk (I'm using 1.8.25.0) i.e. first check for gsm, if doesn't exits then check for slin? It should pick whichever source format requires the least cpu to transcode into the desired output format. So generally that means if there is a source available in the same format as the output then it will use it otherwise it will use slin etc... -- _ -- 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] mixmonitor extension
On 23/01/14 15:21, Marek Cervenka wrote: hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? core show file formats will give you a list of formats your system supports together with the filename extension. Not all may be supported for writing (mp3 being one example I believe). core show file formats Format Name Extensions -- -- slin mp3mp3 h264 h264 h264 g729 g729 g729 g719 g719 g719 gsmgsmgsm g726 g726-16g726-16 g726 g726-24g726-24 g726 g726-32g726-32 g726 g726-40g726-40 h263 h263 h263 gsmwav49 WAV|wav49 g722 g722 g722 ulaw au au alaw alaw alaw|al|alw ulaw pcmpcm|ulaw|ul|mu|ulw siren14siren14siren14 siren7 siren7 siren7 slin192sln192 sln192 slin96 sln96 sln96 slin48 sln48 sln48 slin44 sln44 sln44 slin32 sln32 sln32 slin24 sln24 sln24 slin16 sln16 sln16 slin12 sln12 sln12 slin slnsln|raw slin16 wav16 wav16 slin wavwav g723 g723sf g723|g723sf ilbc iLBC ilbc 30 file formats registered. -- _ -- 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] Weird issue with Set(CALLERID(name)=string);
On 16/01/14 10:47, Tiago Geada wrote:
Hi folks,
We've been having a weird issue... It is happening more often in the
last few months...
Most inbound calls, we have in our dialplan before Queue():
Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});
So when the call rings a member, softphone will show this string
The issue is that sometimes the string showing in the softphone is not
the same. Its a string from a past call, in the latest case I've seen,
from about 40 days ago!!
User took a screenshot, I've searched for that uniqueid showing in
softphone in cdr, and that string was valid for a different call 40
days ago!!
I searched full log, and Set() sets the correct string... I can't
figure why softphone shows a string from a past call !!
:(
Any hints ?
I would leave tcpdump running capturing port 5060 so you can load it
onto wireshark and have a look at the sip headers. That will tell you if
the SIP is incorrect or if its a problem with the client.
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);
Very little as the amount of data being captured is quite small. We have
it running on our production servers which routinely handle a couple of
hundred concurrent calls.
This is the script we use to start off the capture. It uses rolling
capture files so we will always have the last X number of capture logs.
It works very well and we have a custom system which enables us to
search for calls and request traces for them for when we have to
diagnose problems.
#!/bin/bash
cd /var/lib/asterisk/siptraces
DATE=`date +%Y%m%d%H%M%S`
TRACEFILE=/var/lib/asterisk/siptraces/$DATE-siptrace.pcap
nohup /usr/sbin/tcpdump -p -i eth0 -s 0 port 5060 -w $TRACEFILE -C 10 -W
500
On 16/01/14 14:27, Tiago Geada wrote:
You're right, seems like a nice way to debug. Regarding that, how
would the impact be affected running it on asterisk box? I guess only
port 5060 is not too bad
On 16 January 2014 14:09, Gareth Blades
mailinglist+aster...@dns99.co.uk
mailto:mailinglist+aster...@dns99.co.uk wrote:
On 16/01/14 10:47, Tiago Geada wrote:
Hi folks,
We've been having a weird issue... It is happening more often in
the last few months...
Most inbound calls, we have in our dialplan before Queue():
Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});
So when the call rings a member, softphone will show this string
The issue is that sometimes the string showing in the softphone
is not the same. Its a string from a past call, in the latest
case I've seen, from about 40 days ago!!
User took a screenshot, I've searched for that uniqueid showing
in softphone in cdr, and that string was valid for a different
call 40 days ago!!
I searched full log, and Set() sets the correct string... I can't
figure why softphone shows a string from a past call !!
:(
Any hints ?
I would leave tcpdump running capturing port 5060 so you can load
it onto wireshark and have a look at the sip headers. That will
tell you if the SIP is incorrect or if its a problem with the client.
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);
The SIP trace will give you an idea is perhaps something is becoming
corrupted. If you keep a log of the asterisk console output (asterisk
-rvvv) then you will see what it attempts to set the callerid to and any
errors.
Another tip. When you have a look at the sip trace you will see the
call-id. If you make a note of this and run the following replacing the
call-id and the trace file with the appropriate values it will display
the sip trace in a very nice human readable format. tshark comes with
the wireshark pakage and ngrep is part of epel repository if you are
running centos.
tshark -t ad -r '$tracefile' -R 'sip.Call-ID contains $callID' -w - |
ngrep -I - -W byline -t
On 16/01/14 14:57, Tiago Geada wrote:
Second thought, that would only allow me to know if there is a problem
on asterisk or softphone.
Because the old callerid(name) that was presented on the softphone,
belonged to a call made to a different peer, I doubt that it would be
a softphone problem.
Our softphones are fixed with the same peer/extension .. if the wrong
callerid were originally called to the same peer.. I guess that would
be worth it..
even so, I will try and measure the impact on performance, however if
asterisk did send the wrong string, how could I debug that??
On 16 January 2014 14:27, Tiago Geada tiago.ge...@gmail.com
mailto:tiago.ge...@gmail.com wrote:
You're right, seems like a nice way to debug. Regarding that, how
would the impact be affected running it on asterisk box? I guess
only port 5060 is not too bad
On 16 January 2014 14:09, Gareth Blades
mailinglist+aster...@dns99.co.uk
mailto:mailinglist+aster...@dns99.co.uk wrote:
On 16/01/14 10:47, Tiago Geada wrote:
Hi folks,
We've been having a weird issue... It is happening more often
in the last few months...
Most inbound calls, we have in our dialplan before Queue():
Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});
So when the call rings a member, softphone will show this
string
The issue is that sometimes the string showing in the
softphone is not the same. Its a string from a past call, in
the latest case I've seen, from about 40 days ago!!
User took a screenshot, I've searched for that uniqueid
showing in softphone in cdr, and that string was valid for a
different call 40 days ago!!
I searched full log, and Set() sets the correct string... I
can't figure why softphone shows a string from a past call !!
:(
Any hints ?
I would leave tcpdump running capturing port 5060 so you can
load it onto wireshark and have a look at the sip headers.
That will tell you if the SIP is incorrect or if its a problem
with the client.
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);
On 16/01/14 15:29, Kevin Larsen wrote: Not to derail the conversation, Gareth, but do you leave this running full time on your asterisk boxes or just turn it on when you are trying to track problems? On average, how far back can you go for looking at problems? Its normally running full time so if someone reports a problem with a call we can look at the logs and find out exactly what happened. We keep asterisk verbose logs for 3 months, sip traces currently for about a month, and uk-isup traces for a couple of weeks. Most carriers will do something similar. BT for example keep all of their SS7 signalling for 48 hours. Regards Gareth -- _ -- 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] No compatible codecs, not accepting this offer!
On 15/01/14 09:39, Francesco Namuri wrote: Hello James, thanks for your answer, I supposed this too, but my provider answered me that as m=audio 43718 RTP/AVP 8 18 3 101 ^ ^ ^ GSM proposal ^ ^--- G729 proposal ^-- aLaw proposal And that a=rtpmap:18 G729/8000 proposed as media conversion a=rtpmap:3 GSM/8000/1 because the call is made by a mobile I would agree with what your service provider has said. If you look at the RFC http://tools.ietf.org/html/rfc4566#section-5.14 the '8 18 3 101' parameters are a list of media formats. The first is the one which should be used but (preferred choice) but the other may be used. Numbers in the range 96-127 are dynamic payload types and these must have a corresponding 'a=' line specifying the payload type and the codec options. Lower numbers have static payload assignments and according to that RFC dont have to have corresponding 'a=' lines. A list of types can be found at http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml However in all SIP traces I have seen there has always been a 'a=' line for every payload type offered. The static payload type numbers are used but there is still the 'a=' line. -- _ -- 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] Maximum number of users
You need to decide which codecs you are going to allow to be used on the SIP side. As you are connecting to E1 then the standard codec would be g711 alaw or ulaw. You could force the SIP side to use the same codec but it uses about 100Kbps of bandwidth so quite a bit higher than other codecs and some customers may prefer to use a higher compression codec so they dont have to have a faster internet connection. If you start allowing other codecs then asterisk will need to perform codec translation which consumes CPU. You will be limited by the E1 cards you can fit into asterisk. Generally its best to keep to a maximum of 2 PCI cards in a server as they each generate a lot of interrupts. A couple of Sangoma A116 will give you 32 E1 circuits which can handle 960 calls. That may well be pushing the server to its limits of what it can do even if it is just handling basic calls. So you will want 3 of these systems and preferably four as you will want some redundancy. You would want something like opensips at the front end on a couple of redundant servers. You can use the load balancing module in opensips to spread the calls evenly between the available asterisk servers. You should also have a think about whether you wish to use asterisk at all. Opensips will do the load balancing that you need and can perform the billing actions aswell. If all asterisk is doing is performing sip to E1 conversion there may be hardware solutions which will be more cost effective and require less maintenance. On 20/12/13 10:08, bilal ghayyad wrote: Thanks a lot for the help from the all. Without using PBX functionalities (like conference or pickup and so on), only basic calls. So how many concurrent calls can support? The idea is, we need to use Asterisk with ISP which will be service provider for sip calls for the subscribers, and the asterisk should be connected with E1s to do calls within the country. The registered users will reach up to 100 000 users and the concurrent calls will reach up to 2000 or 3000 calls. Appreciate the kindly advise. Regards Bilal On Wednesday, December 18, 2013 6:10 PM, Tech Support aster...@voipbusiness.us wrote: You can have tens of thousands of phones as long as no one makes or receives any calls J. The better question to ask is how many concurrent calls have people been able to make. The quick answer is it depends on many things. John *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *bilal ghayyad *Sent:* Wednesday, December 18, 2013 9:46 AM *To:* Asterisk Users Mailing List - Non- Commercial Discussion *Subject:* [asterisk-users] Maximum number of users Hello; Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later? Regards Bilal -- _ -- 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] Maximum number of users
Thats fine for calculating how many users a particular speed network
connection can cope with. 640 concurrent calls on a 100Mbps connection
is doable on a decent machine as long as you are not doing much codec
translation.
Once you get to the point where you start having hundreds of users then
you really need to think about redundancy. OpenSIPS is a good way
forward as it is very light weight and can easily handle tens of
thousands of registered phones. As calls come in then it can forward
them directly or if additional services are required such as possibly
voicemail it can pass them onto one or more asterisk servers in a load
balancing arrangement. Using a couple of database servers replicating to
each other and a virtual IP address (or DNS SRV if the clients properly
support it) will give you redundancy and allow you to add more server to
expand the capacity.
On 19/12/13 05:19, Brian LaVallee wrote:
Hi Bilal,
Assuming you have the latest hardware, sufficient memory, cpu, etc...
The key to determine the maximum number of users comes down to the
office type, RTP path, network interface, and primary codec used.
First we need to determine the over-subscription rate, how many people
will be using the phones at any given time.
For a call center, the ratio is 1:1.
For a normal office, the industry standard is 4:1.
{This ratio is also used to determine the number of PSTN channels you
will need too}
Will the PSTN connections be Digium card(s) in your server or external
gateway(s)?
Assuming Diguim card(s), the RTP will be going through your server.
Determine the network interface. 10/100/1000baseT
Then we need to consider the largest codec used, and divide the
available bandwidth by the typical packet size.
µ-law/A-law is roughly 80 kbps, so we can support 128/1280/13107 audio
streams.
Divide that by 2 (just to be safe) and allow RTP in both directions.
64/640/6553
Now multiple the result by the over-subscription ratio. 4:1 =
256/2560/26212
So we see that the maximum number of users is 2560 for a normal office
when there is a 100baseT NIC in your Asterisk server.
You would also need to have 640 channels (28 T1 PRI's) connecting to
the PSTN.
/Using SIP trunks to connect to the PSTN through the same 100baseT NIC
will reduce the maximum number of users you can support./
The real challenge is not supporting thousands of users (IP Phones),
it's connecting a sufficient number of PSTN connections to support
those users.
Sincerely,
Brian LaVallee
On 12/18/13, 11:45 PM, bilal ghayyad wrote:
Hello;
Can someone advise me what is the maximum number of users (IP Phones)
that can be supported by asterisk 1.8 or later?
Regards
Bilal
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Re: [asterisk-users] Maximum number of users
We have a machine with a quad core 'Intel(R) Xeon(R) CPU E5-1410 0 @ 2.80GHz' running asterisk 11.2-cert with ingress and egress all sip. Fastagi running as a daemon (written in perl) performing cdr updates at call start, answer and call end together with a query when a call comes in to get information on what to do with it. Local Mysql 5.5 database. With all this we easily handled 450 simultaneous calls with many of them also performing call recording. I think we typically had 60% idle on each processor so the box could be pushed a lot more. On 18/12/13 15:46, Tech Support wrote: How did the system behave with 244 calls? I've been able to make 1,024 concurrent faxes (which tend to use more resources than audio calls) in the lab. The problem I had was after the faxes were transmitted, things couldn't keep up and kept dumping core. Two things were going on, (1) the CDR was written to MySQL and (2) a FastAGI script (I use the AGISpeedy PERL package) to write a log entry also to MySQL. I tried switching the CDR's to sqlite and that seemed able to keep up, except that its concurrency issues were a problem. If MySQL is the problem, I could probably optimize it better, but it doesn't explain the Asterisk core dumps. It might be related to the number of FastAGI scripts running, I'm not sure at this point. Regards; John *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Keith Sloan *Sent:* Wednesday, December 18, 2013 10:10 AM *To:* bilal ghayyad; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Maximum number of users The better question is, maximum number of users (IP Phones) can your hardware support. I have * deployments with 300-600 phones. - works fine. though concurrent calls has never seen more then 244. Also at this point I have to ask, for this to be any concern to you, you must either A, Make tons of internal calls. OR B have multiple T1's/lots of sip channels? Regards, Keith Sloan Voice Operations Center Vianet 705-222-9996 X7203 1-800-788-0363 X7203 kei...@vianet.ca mailto:kei...@vianet.ca On Wed, Dec 18, 2013 at 9:45 AM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: Hello; Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later? Regards Bilal -- _ -- 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] Asterisk RTP Questions
On 27/11/13 14:12, James Bensley wrote: What is the maximum delay RTP will tolerate one way (Does Asterisk have a limit too)? Can this be tuned (increased or decreased) within Asterisk (I'm thinking of DSL customers where we may have this issue between our PBXs and the customer)? There isnt one really. There is a rtptimeout setting but that is designed to hang up a call if no rtp has been received for X seconds. Its going to normally be something long like 30 seconds as is designed to end a call if the sip endpoint you were talking to dies. How can I monitor for such an effect? Does anyone else have any / or had any issue like this? What direction does the audio stop? have you looked at the SIP traces to see if there are and reinvites at the same time? You havent said if your server is directly connected to the internet with its own public IP address or whether it goes via NAT. Kind regards, James. -- _ -- 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] Ast11: How to see call progress like in Ast = 1.8
On 19/11/13 16:44, Bas Rijniersce wrote: Hi, I just did a test install of Ast 11, and have trouble getting the same logging information that Ast 1.x provided. I'm looking specifically for the logging around call progress / dialplan actions. In ASt 11 I've done the same thing that I did before: core set verbose 60 I also tried overwriting the logger.conf with the distribution one from Ast 11, and setting option logger set level verbose on (never did that on older versions, but was wondering if that would make a difference). Still no joy, Googling around for an answer I did see a changelog with an example of the Call Identifier that shows a detailed logline (of level verbose, something I don't get in 11). ButI've been unable to find an answer. Any hints/tips, I must be overlooking something basic.. TX! Bas When we upgraded it still just worked as usual. The only thing we noticed is when connecting to asterisk 'asterisk -r' by default didnt show the same log level. Using 'asterisk -rvvv' solved it. The verbose logs stored in /var/log/asterisk were the same though apart from the additional call-id entry. In our logger.conf the only significant line appears to be :- messages = notice,warning,error,verbose -- _ -- 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] Unix connections not always disconnecting
On 07/11/13 11:20, Ishfaq Malik wrote: Hi We are using asterisk 1.8.23.1 We have a script that runs on a minute cron which polls the asterisk server for 3 bits of information by using asterisk -rx 'command' which then gets pushed to a graphite server we have 99% of this runs smoothly. Out of interest what are you trying to monitor? We tend to use cacti for graphing and snmp provides all the information we require. -- _ -- 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] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9
On 11/10/13 18:43, Tiago Geada wrote: Hi, Seems a great workaround from Gareth Blades. Thanks I will try it. Any way to make asterisk log a line in /var/log/messages ? I normally have all the verbose output sent to the log file so anything in the NoOp() line gets logged to the file so thats what I use. You could use the Log() or Verbose() applications if you only have errors written to the file as with those commands you can specify a log level. -- _ -- 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] Capture Media IP in CDR (CDR)
On 13/10/13 20:06, CDR wrote:
I am quite surprised about the degree of surprise in the group. A few
days ago, somebody called a school and issued a threat, through my
network. The call came from China, but of course it was US caller. The
DA wants to know where call came from. The caller ID is Restricted
and the chinese carrier is playing games.
How do you think it works with regular telecomms?
The police need to follow the trail. All you need to provide is that the
call came in via carrier X and they will then go onto that carrier to
see where the call originated.
My advice would be to :-
1) Add ${SIPCALLID} to your cdr records. This is the unique ID for the
sip call which can be used later.
2) Run tcpdump -p -s 0 port 5060 -w $siptrace.pcap -C 10 -W 500
-C is how big the dump will be and -W is how many capture files to get
before overwriting the old one. make the -C value (10 in this case) big
enough so each file lasts 15 minutes or so and the '-W' value big enough
so you keep however many days records you need.
3) Now when you get a request look in the cdr records for the callid.
Assuming for example its qwertyuiop then look at the time and pick the
pcap file covering that time range. Make sure you have the 'wireshark'
and 'ngrep' linux packages installed. Then :-
tshark -t ad -r TRACEFILE -R 'sip.Call-ID contains qwertyuiop' -w - |
ngrep -I - -W byline -t
The standard output now contains a complete sip trace and you will be
able to see all the media endpoints and exact timings.
Thats basically what we do for getting call diagnostics.
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Re: [asterisk-users] Asterisk 11 sending comfort Noise
On 08/10/13 17:02, Eric Wieling wrote: I have an Asterisk 1.4 box which is sometimes getting the message below. Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. 209.220.119.19 is an Asterisk 11 box. [Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.220.119.19 It might be that the other asterisk box has remotely bridged the call so its not storing and forwarding the RTP packet which contains the silence suppression information. -- _ -- 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] Dahdi_dummy is more accurate than core timer?
On 02/10/13 12:17, Johan Wilfer wrote: Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I re-ran these tests on another server showing the same thing. - Can anybody comment on why DAHDI with core timers drop down to 99.6% occasionally? - Is a hardware-card for timing the most efficient way to get timing even if I just use the card for the timing? Its a little different when you are using meetme as its an application built into dahdi itself and not a native asterisk application. It will therefore always use dahdi for its timing. If dahdi doesnt have a hardware interface (sangoma sell a usb based timing source if you want a hardware source) then it will use a software timing source of some form. I dont know what method it uses. Asterisk itself will use dahdi for timing but if res_timerfd is available it will use that itself. With your kernel version timerfd should be available. So if you run the following command asterisk will perform 1024 ticks over 1000ms which is equivalent to the dahdi test of 8192 over 8000ms (if you run it a few times). You can see in my case asterisk is using timerfd and no issues at all with the timing. timing test 1024 Attempting to test a timer with 1024 ticks per second. Using the 'timerfd' timing module for this test. It has been 1000 milliseconds, and we got 1024 timer ticks We are developing a conferencing feature and rather than use meetme with its dahdi requirement we are using confbridge10 instead so we dont have to have dahdi installed and it will use the better timerfd timing source. More information about timing sources can be found at :- https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces -- _ -- 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] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9
On 02/10/13 16:13, gincantalupo wrote: Hi Garet, ok but since the messages contain my own public IP with this method I'm banning my public IP not the real attacker IP. Am I wrong? Giorgio No the asterisk dialplan entry is pulling the IP address out of the SIP Contact: header which in the attacks we have seen always seems to be the correct IP address. -- _ -- 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] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com
wrote:
Hi,
I get a lot of these messages on my Asterisk CLI:
Failed to authenticate user
1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9
as if my PBX machine is trying to authenticate to itself. It seems
someone is attacking my asterisk PBX.
Is there a way to fix this problem?
in sip.conf I have guest connections permitted and have them going to
the default context which contains :-
[default]
; all unauthenticated connection attempts from the internet come in here.
exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
${SIP_HEADER(Contact)})
exten = _[+*#0-9].,n,Congestion
Then in fail2ban I have it match the following :-
failregex = Registration from .* failed for \'HOST\' - Wrong password
Unauthenticated call attempt .*\@HOST\:
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Re: [asterisk-users] Generating a different countries ringtone on a per call basis
On 26/09/13 16:43, Rusty Newton wrote: On Thu, Sep 26, 2013 at 10:08 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 26/09/13 14:59, Rusty Newton wrote: Try the following: extension = 6001,1,Set(CHANNEL(tonezone)=us) same = n,Dial(SIP/6001,,r(ring)) The argument passed to the r option should be the specific tone in the category of the tonezone you are setting. Thanks. I did try that as pretty much the first thing I tried but it continued to play the UK ring tone. Its not a big issue as we can work around it by playing music on hold instead which is a recording of the required ring tone. Having asterisk generate it just seemed the neater option. Are you sure you specified an argument to the 'r' option? Or did you just try 'r' without an argument? For me.. if I specify a uk tonezone, to get it playing uk tones I have to specify an argument to the 'r' option. If I try just 'r' by itself then I get US tones. You would think, that without specifying an argument, it should default to the tonezone in use on the channel. That may be a bug or oversight. What version of Asterisk were you using, and what channel type? That could well be it. It would have been with the standard 'r' option and not 'r(ring)' as thats the way our feature is currently programmed. I have just tested it with r(ring) and that works. Thanks -- _ -- 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] Is this SDP payload Asterisk created valid?
We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails. We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network. After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488 response from the far end. This packet seems to be updating the version for the o= session id which is fair enough. Its sending the c= connection data but not the m=audio line which appears to be what the remote end is complaining about. Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? Thanks Gareth U 2013/09/27 11:04:55.352854 88.x.x.25:5060 - 213.x.x.24:5060 INVITE sip:0844xx@146.x.x.10:54900 SIP/2.0. Via: SIP/2.0/UDP 88.x.x.25:5060;branch=z9hG4bK62215713. Route:sip:213.x.x.24;lr=on;ftag=as691af817;did=ecd.c2dc96e6. Max-Forwards: 70. From:sip:01628xx@88.x.x.25;tag=as691af817. To:sip:0844xxx...@freespeech.co.uk;tag=ee7a6c7cad57f096i1. Contact:sip:01628xx@88.x.x.25:5060. Call-ID: 2eeb643d086234de59a1fd4e78170d3f@88.x.x.25:5060. CSeq: 104 INVITE. User-Agent: Asterisk PBX 11.2-cert2. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 110. . v=0. o=root 716216031 716216033 IN IP4 88.x.x.35. s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.35. t=0 0. # U 2013/09/27 11:04:55.365458 213.x.x.24:5060 - 88.x.x.25:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 88.x.x.25:5060;branch=z9hG4bK62215713;rport=5060. From:sip:01628xx@88.x.x.25;tag=as691af817. To:sip:0844xxx...@freespeech.co.uk;tag=ee7a6c7cad57f096i1. Call-ID: 2eeb643d086234de59a1fd4e78170d3f@88.x.x.25:5060. CSeq: 104 INVITE. Server: OpenSIPS (1.5.3-notls (x86_64/linux)). Content-Length: 0. . # U 2013/09/27 11:04:55.431674 213.x.x.24:5060 - 88.x.x.25:5060 SIP/2.0 488 Not Acceptable Here. To:sip:0844xxx...@freespeech.co.uk;tag=ee7a6c7cad57f096i1. From:sip:01628xx@88.x.x.25;tag=as691af817. Call-ID: 2eeb643d086234de59a1fd4e78170d3f@88.x.x.25:5060. CSeq: 104 INVITE. Via: SIP/2.0/UDP 88.x.x.25:5060;rport=5060;received=88.x.x.25;branch=z9hG4bK62215713. Record-Route:sip:213.x.x.24;lr=on;ftag=as691af817. Contact: freespeechsip:0844xx@146.x.x.10:54900. Warning: 304 spa Media type not available. Server: Cisco/SPA303-7.5.4. Content-Length: 0. -- _ -- 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] Is this SDP payload Asterisk created valid?
On 27/09/13 14:15, Gareth Blades wrote: Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? Reading RFC2327 it cais the c= line 'must' be present in all updates while 'm=' media lines are optional. I am therefore inclined to believe that Asterisk is working correctly and there is a bug in the customers SIP equipment. But thats just my personal interpretation from briefly reading the standard. -- _ -- 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] Is this SDP payload Asterisk created valid?
On 27/09/13 14:36, Joshua Colp wrote: Gareth Blades wrote: We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails. We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network. After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488 response from the far end. This packet seems to be updating the version for the o= session id which is fair enough. Its sending the c= connection data but not the m=audio line which appears to be what the remote end is complaining about. Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? The SDP you posted should be fine BUT my question becomes... have you modified chan_sip at all? I don't think it should be possible for it to not put any media lines in... Cheers, No we havent made any changes to chan_sip. The servers were a fresh install a short while ago straight to 11.2-cert1 as we wanted a later kernel version to make use of the new timing source it provides. We then upgraded to cert2 after it was released. The only thing we have changed is the setting of a DYNAMIC_FEATURES variable which was stopping remote bridging from being performed which is probably what has highlighted this fault. Thanks Gareth -- _ -- 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] Is this SDP payload Asterisk created valid?
On 27/09/13 14:28, Gareth Blades wrote: On 27/09/13 14:15, Gareth Blades wrote: Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? Reading RFC2327 it cais the c= line 'must' be present in all updates while 'm=' media lines are optional. I am therefore inclined to believe that Asterisk is working correctly and there is a bug in the customers SIP equipment. But thats just my personal interpretation from briefly reading the standard. Well looking at RFC4566 section 5 :- Some lines in each description are REQUIRED and some are OPTIONAL, but all MUST appear in exactly the order given here (the fixed order greatly enhances error detection and allows for a simple parser). OPTIONAL items are marked with a *. Session description v= (protocol version) o= (originator and session identifier) s= (session name) i=* (session information) u=* (URI of description) e=* (email address) p=* (phone number) c=* (connection information -- not required if included in all media) b=* (zero or more bandwidth information lines) One or more time descriptions (t= and r= lines; see below) z=* (time zone adjustments) k=* (encryption key) a=* (zero or more session attribute lines) Zero or more media descriptions Time description t= (time the session is active) r=* (zero or more repeat times) Media description, if present m= (media name and transport address) i=* (media title) c=* (connection information -- optional if included at session level) b=* (zero or more bandwidth information lines) k=* (encryption key) a=* (zero or more media attribute lines) So if I am reading that correctly the m= line is required only if we include a media description entry. It basically sais its required if we decide to include it (yes I know that doesnt make sense). What I presume this means is that if a media description is included such as there being a 'a=' line then the 'm=' line then becomes required. So it still sounds like Asterisk is behaving correctly. -- _ -- 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] Realtime Mysql
On 27/09/13 17:47, Phibee Network Operation Center wrote: Hello, I am looking to know if it is possible to modify the SQL query that is on Realtime sip accounts. I want multiple servers use the same sql table, so getting an extra server field to indicate that the data is valid on the X server is this possible? thank you in advance jerome I dont know but you should be able to at least create multiple mysql views with each one being referenced by the appropiate server. -- _ -- 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] Generating a different countries ringtone on a per call basis
On 26/09/13 14:59, Rusty Newton wrote: On Wed, Sep 25, 2013 at 6:45 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: We can use the Dial() command with the 'r' option in order to generate the UK ringtone (as we are UK based the default is UK). How do we generate a USA ringtone for example? I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us' (and calling Progress() beforehand) and although this works for Playtones() the Dial command still continues to play the UK ringtone. Any ideas? Try the following: extension = 6001,1,Set(CHANNEL(tonezone)=us) same = n,Dial(SIP/6001,,r(ring)) The argument passed to the r option should be the specific tone in the category of the tonezone you are setting. Thanks. I did try that as pretty much the first thing I tried but it continued to play the UK ring tone. Its not a big issue as we can work around it by playing music on hold instead which is a recording of the required ring tone. Having asterisk generate it just seemed the neater option. -- _ -- 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] Problems sending log to rsyslog
On 26/09/13 15:25, Mauricio Tavares wrote: So I have asterisk 1.8.23 and want to send my logs to rsyslog. I tell asterisk to use syslog in addition to messages: root@voip:~# tail -10 /etc/asterisk/logger.conf ;debug = debug console = notice,warning,error ;console = notice,warning,error,debug messages = notice,warning,error ;full = notice,warning,error,debug,verbose,dtmf,fax ;syslog keyword : This special keyword logs to syslog facility ; syslog.local0 = notice,warning,error ; root@voip:~# After reloading (asterisk -rx 'logger reload') the logger, it seems that Asterisk is happy: root@voip:~# asterisk -rx 'logger show channels' Channel Type StatusConfiguration --- --- syslog.local0 Syslog Enabled- NOTICE WARNING ERROR /var/log/asterisk/messages File Enabled- NOTICE WARNING ERROR Console Enabled- NOTICE WARNING ERROR root@voip:~# So I set rsyslog: root@voip:~# fgrep asterisk /etc/rsyslog.d/50-default.conf local0.* /var/log/asterisk/messages.log root@voip:~# and restart it. And then check the asterisk log directory: root@voip:~# ls -lh /var/log/asterisk/ total 3.7M drwxr-xr-x 2 asterisk asterisk 4.0K Jul 22 20:57 cdr-csv drwxr-xr-x 2 asterisk asterisk 4.0K Jun 28 14:16 cdr-custom -rw-rw 1 asterisk asterisk 252K Sep 26 09:37 messages -rw-rw 1 asterisk asterisk 248K Sep 22 05:14 messages.1 -rw-r- 1 syslog adm 0 Sep 26 06:47 messages.log -rw-rw 1 asterisk asterisk 118 Sep 26 10:07 queue_log root@voip:~# It does not seem like much is being written to messages.log compared to messages. Anything I missed? Have you checked the /var/log/asterisk directory permissions? I dont know how rsyslog is setup on your system but its possible it gets started as root, sees the destination file doesnt exist so creates it and sets the file permissions, and then drops down to running as the syslog user. At this point it doesnt have write permission to the /var/log/asterisk directory so cannot append to the file. -- _ -- 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] users can not hear the audio playback sometimes
On 25/09/13 09:54, Kumar Shantanu wrote: I am facing a strange problem on my asterisk box (using isdn lines with pri card installed on it). Normal incoming/outgoing calls are working perfectly fine. When a user dial a wrong/out-of-service number they don't hear back any such message like The number is wrong or user is switched off in some cases, and it's just a silence for the user. Now while troubleshooting we set recording to ON for the extension so we can see what's there in the recording file(.gsm). When I am listing to the recording files it contains everything which users should be listening actually, like the number you are dialling is not correct. So the question is why user is not able to listen to this while asterisk itself is getting the message perfectly fine from the provider ? Am I missing something my dahdhi.conf ? Or anybody having any idea about this can please put me through the right direction. Try calling Progress() just before the dial command. Without this Asterisk wont send the SIP/183 Session Progress and send the inband audio until the call is answered. -- _ -- 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] users can not hear the audio playback sometimes
On 25/09/13 11:21, Kumar Shantanu wrote: Thanks Gareth , Try calling Progress() just before the dial command. Without this Asterisk wont send the SIP/183 Session Progress and send the inband audio until the call is answered. Do I need to change something in asterisk dial plan ? I am using freepbx to mange asterisk graphically. Yes you will see a section called [macro-dialout-trunk] Within that there will be a line line :- exten = s,n,Dial(something) Just before that line add :- exten = s,n,Progress() You will then need to reload the dialplan (dialplan reload from the asterisk prompt) and you can give it a go. -- _ -- 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] Generating a different countries ringtone on a per call basis
We can use the Dial() command with the 'r' option in order to generate the UK ringtone (as we are UK based the default is UK). How do we generate a USA ringtone for example? I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us' (and calling Progress() beforehand) and although this works for Playtones() the Dial command still continues to play the UK ringtone. Any ideas? Thanks Gareth -- _ -- 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] users can not hear the audio playback sometimes
On 25/09/13 13:57, Kumar Shantanu wrote: Thank you Gareth, It worked like a charm. The only problem I am having is now, when I do some changes in my freepbx and reload it just rewrites my dial play , I will try to fix it though. Thanks again I did see in the console output it doing a GotoIf() and checking if a custom trunk was defined just before the original dial command. Maybe there is a way you can define a custom trunk and basically copy the standard part of the dialplan but with the extra line you added? Sorry I dont know the product so cant tell you how to do 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
Re: [asterisk-users] Strange Error
On 25/09/13 15:42, Andrew Colin wrote: Hi Guys, Anyone ever seen this before. on asterisk 1.8 if i set one of my pabx extensions to show private number and send a call over VoIP with g729 the call fails but with alaw it works. If i enable the callerid on g729 it also works see error below From: Anonymoussip:anonymous@anonymous.invalid;user=phone;tag=07d44838 That single line doesnt really help. You would need to give a full sip trace of a working withheld alaw can and a working and non working g729 call. Also the asterisk console output for the failed call. -- _ -- 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] RTP not being switched between both SIP endpoints
On 18/09/13 12:40, Kenny Watson wrote: Hi, Since opensips is not handling media (i presume) is the audio not already going direct from asterisk to the other endpoint? Thanks Kenny Opensips wasnt handling the media so the audio was between the original caller and asterisk (with the signalling being relayed by opensips). It was just when we dialled onto the final destination via SIP asterisk stayed in the loop and didnt issue a reinvite. Its all fixed now. Although we weren't using any features the AGI application was setting DYNAMIC_FEATURES to an empty string which was enough to keep asterisk in a loop. We stopped the AGI from setting the variable if there were no features and it started working. Thanks Gareth -- _ -- 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] RTP not being switched between both SIP endpoints
We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call coming from opensips. Directmedia is not specifically defined so its using the asterisk default value. * Name : vmpubopensips3 Description : Secret : Not set MD5Secret: Not set Remote Secret: Not set Context : from-pubopensips Record On feature : automon Record Off feature : automon Subscr.Cont. : Not set Language : Tonezone : Not set AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : no Force rport : Auto (No) Symmetric RTP: No ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 88.x.x.x Addr-IP : 88.x.x.x:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : (gsm|ulaw|alaw) Codec Order : (alaw:20,ulaw:20,gsm:20) Auto-Framing : No Status : Unmonitored Useragent: Reg. Contact : Qualify Freq : 6 ms Keepalive: 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No When the call comes in the SDP contains :- v=0. o=root 973184584 973184584 IN IP4 81.x.x.x s=session. c=IN IP4 81.x.x.x t=0 0. m=audio 11370 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and we reply back with :- v=0. o=root 822402971 822402971 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10428 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. When we send the outbound SIP information we advertise the following SDP :- v=0. o=root 431105643 431105643 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10144 RTP/AVP 8 3 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and the other end replies with :- v=0. o=hksbc1a 609621538 609621538 IN IP4 203.x.x.x s=sip call. c=IN IP4 203.x.x.x t=0 0. m=audio 34146 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. In the Dial() command the only option we are using is M() which is used to run a macro when the call is answered. This is used to update cdr records and perform other features if they are enabled. In this case we are just updating the cdr records so I would expect the audio to be switched as soon as the macro finishes. Any ideas what could be wrong? We are running Asterisk PBX 11.2-cert2 Thanks Gareth -- _ -- 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] executing the h extension at the real hangup of the call
On 13/09/13 12:31, Henrik Westerberg wrote:
Hi,
I am running Asterisk 11.3 with both SIP and ISDN. When dialing out
(always over SIP) I want to keep track of who answered and of the
length of the call.
[outgoing-dev2]
exten = h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished)
exten = _X.,1,NoOp(Will send call to ${CC_DIALSTRING})
exten = _X.,n,Dial(${CC_DIALSTRING}, 60,
M(uploadpeer-dev2^${CC_CALLID})em)
exten =
_X.,n,Agi(agi://localhost/ajpbxtest.agi?status=faileddialstatus=${DIALSTATUS})
The h extension is called correctly when the call comes in over IP and
when I record the call. But when the call has come in over SIP the h
extension is called directly after the call is answered so all the
call gets length 0 in my own database.
I guess that I could record the calls and throw away the recordings
afterwards. In this way the RTP would stay on the server. But is there
not a cleaner way to get Asterisk to execute the h extension (or
another possibility to fix a callback somewhere) when the the
Disconnect comes in over SIP?
I have no idea why you are seeing the h extension being run before the
call ends. Its not something I have ever seen happen.
Whether or not Asterisk stays in the RTP media path makes no difference
as it will always stay in the SIP signalling path and its that which
controls the call establishment and termination.
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Re: [asterisk-users] Use SRV for failover proxy
On 06/09/13 09:42, Dominique Haeber wrote:
Hi all,
is it possible that asterisk uses two proxies with SRV?
The enddevices are registered on one of the two Proxies (Kamailio).
The two proxies communicate with each other.
And asterisk can choose one of this proxies with SRV.
asterisk
| \
|\
Proxy1Proxy2
I have tries to solve this problem with two trunks for this proxies
and Dial(...@proxytrunk) but on this way the properties of the
enddevices are lost and the properties from trunk are taken.
So i'd like to use SRV, if that is possible.
Very thanks
Dominique
No asterisk will always use the first SRV record and wont load balance
or switch to a backup if its not reachable.
What we do is have each endpoint defined in sip.conf with qualify=yes
and then in the dialplan use the ${SIPPEER(x)} variable to pull out the
status of each peer and pass it into an AGI application to perform the
load balancing etc...
If you are happy with wone being a primary and one being a backup then
if you have qualify=yes set for both you could just dial using the first
one and then an execif hangupcause=20 then try dialing the backup.
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Re: [asterisk-users] asterisk and encryption
On 02/09/13 08:17, gpxctawjc...@irational.org wrote: i am running Asterisk 1.6.2.5-0ubuntu1.4 and would like to know how to incorporate [default] encryption can you point me to any guides please ? do i need to upgrade ? many thanks There are two things :- SIP TLS so that the invites are encrypted. This will enable you to encrypt the information about what numbers are being called etc... https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport The other is SRTP so that the actual audio is encrypted :- http://www.voip-info.org/wiki/view/Asterisk+SRTP SRTP is supported natively in Asterisk 1.8 and later so you should definetly upgrade. -- _ -- 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] Asterisk 1.8.15-cert3, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, 11.5.1 Now Available (Security Release)
On 28/08/13 19:34, Rusty Newton wrote: On Wed, Aug 28, 2013 at 11:26 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 27/08/13 19:20, Asterisk Development Team wrote: The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, and 11.5.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases http://www.asterisk.org/downloads/asterisk/all-asterisk-versions sais :- Latest Version - 11.2-cert2 However if you download it its the old cert1 version. Going direct to :- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ there is no cert2 version there yet. It looks good now. Perhaps you were looking at cached versions of those pages? Were you eventually able to get the cert2 versions? I can see it there now fine thanks. Its strange as I also downloaded the changelog.current file which I hadnt downloaded before and even that was the old version. The cert1 tar.gz isnt there any more and as I did end up downloading that version I think it just took a while until the new files were replicated onto the server. Thanks -- _ -- 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] How to reply with 480 Call-limit to incoming SIP call ?
On 29/08/13 14:42, Olivier wrote:
Thanks for your very helpful reply.
1.My system prints out:
CLI core show application Hangup
-= Info about application 'Hangup' =-
[Synopsis]
Hang up the calling channel.
[Description]
This application will hang up the calling channel.
[Syntax]
Hangup([causecode])
[Arguments]
causecode
If a causecode is given the channel's hangup cause will be set
to the given value.
[See Also]
Answer(), Busy(), Congestion()
How could we improve this Arguments section so that other Asterisk
admins can find available causecode values ?
Have a look in the source code in channels/chan_sip.c and you will see :-
const char *hangup_cause2sip(int cause)
{
switch (cause) {
case AST_CAUSE_UNALLOCATED: /* 1 */
case AST_CAUSE_NO_ROUTE_DESTINATION:/* 3 IAX2:
Can't find extension in context */
case AST_CAUSE_NO_ROUTE_TRANSIT_NET:/* 2 */
return 404 Not Found;
case AST_CAUSE_CONGESTION: /* 34 */
case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
return 503 Service Unavailable;
case AST_CAUSE_NO_USER_RESPONSE:/* 18 */
return 408 Request Timeout;
case AST_CAUSE_NO_ANSWER: /* 19 */
case AST_CAUSE_UNREGISTERED:/* 20 */
return 480 Temporarily unavailable;
case AST_CAUSE_CALL_REJECTED: /* 21 */
return 403 Forbidden;
case AST_CAUSE_NUMBER_CHANGED: /* 22 */
return 410 Gone;
case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
return 480 Temporarily unavailable;
case AST_CAUSE_INVALID_NUMBER_FORMAT:
return 484 Address incomplete;
case AST_CAUSE_USER_BUSY:
return 486 Busy here;
case AST_CAUSE_FAILURE:
return 500 Server internal failure;
case AST_CAUSE_FACILITY_REJECTED: /* 29 */
return 501 Not Implemented;
case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
return 503 Service Unavailable;
/* Used in chan_iax2 */
case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
return 502 Bad Gateway;
case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /*
Can't find codec to connect to host */
return 488 Not Acceptable Here;
case AST_CAUSE_INTERWORKING:/* Unspecified
Interworking issues */
return 500 Network error;
case AST_CAUSE_NOTDEFINED:
default:
ast_debug(1, AST hangup cause %d (no match
found in SIP)\n, cause);
return NULL;
}
For any given hangup cause you can change the sip response there. For a
list of the hangup numbers and the internal variable name look in
include/asterisk/causes.h
So if you change chan_sip.c and add the following just before the
'AST_CAUSE_NOTDEFINED' line and recompile and reinstall you should in
theory be able to do a Hangup(44) to achieve what you want.
case AST_CAUSE_REQUESTED_CHAN_UNAVAIL:/* 44 */
return 480 Temporarily Unavailable (Call limit);
Thats only in theory. I havent tested it myself and I am not an asterisk
developer.
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Re: [asterisk-users] Need input on scalable system design...
On 27/08/13 18:16, Gregory Malsack wrote: Hey All, Growing call center. Currently at about 200 call center staff, running about 1000 calls per hour. Gearing up to double that. Not too sure that a single server will support that growth. So, I'm trying to come up with ways to scale the system and still maintain a simplistic design. So I'd like to bounce some ideas around. Currently I am running on a Dell 1950, dual quad core 2.33ghz xeons, with 16gb ram, and 2 tce400p cards. This server is managing the full load of the company. We are recording all calls, running ivr, queues, cdr, cel, and web for reporting. I currently have another 1950 of the exact same specifications as a cold spare. Here's where you can see drawings of my current connectivity and an optional connectivity I'm contemplating... *MailScanner has detected a possible fraud attempt from www.linkedin.com claiming to be* http://www.paydaysupportcenter.com/current.pdf http://www.linkedin.com/redirect?url=http%3A%2F%2Fwww%2Epaydaysupportcenter%2Ecom%2Fcurrent%2Epdfurlhash=qLsB_t=tracking_anet*MailScanner has detected a possible fraud attempt from www.linkedin.com claiming to be* http://www.paydaysupportcenter.com/option.pdf http://www.linkedin.com/redirect?url=http%3A%2F%2Fwww%2Epaydaysupportcenter%2Ecom%2Foption%2Epdfurlhash=CJG1_t=tracking_anet As you can see I currently have a separate sql server and a separate storage server for the call recordings. This is all working fine. However, I'm thinking for scalability I should be looking to migrate to a configuration similar to the one in option.pdf. Where I have a VOIP gateway server that simply relays traffic and possibly can do some load balancing or intellegent routing. But nothing more then that, and possibly a second one of these online as a hot failover. Then have separate sql, storage, (i forgot it in the pic) web, and asterisk servers behind that on separate dedicated network. Here's my dilemma though, how do I balance the load across multiple machines for scalability... Since 95% of our calls come into queues, I need to be able to maintain queue stats and presence across all of the servers. Thus far, I've got everything except the extensions.conf file into the mysql database. I thought about setting up 2 servers, 1 for sales, and 1 for customer service, then possibly break out each call queue to it's own server as things grow. Just not sure if that's the right way to go. Then regarding extensions.conf, I've read that it too can be placed in the sql database and accessed via switch. however it's resource intense, so now I'm thinking of maybe putting that file on the nfs server for all of the boxes to read from. As for the design of that file, I was kind of thinking of a modular design within the file using various goto's and gosubs. Our business model is based on affiliates and corporate marketing, so we have a ton of did's that follow the same call flow with minor modifications in some variables, as well as variations in call flow, and hours of operation. Thus the modular design of the call flow. Then the primary inbound context would simply be a list of did's pointing to a goto with a list of the variations and variables for the did. Ok, now that I've melted your brains thoughts? Thanks all in advance for the discussion... Greg We have a similar server but a single quad core at 3Ghz. It easily handles 400 concurrent calls with a lot shorter average call duration than you have. It doesnt do as much call recording but does do a lot of AGI stuff. With regard to nfs thats fine for non real time stuff. Personally we have a test machine and multiple live machines and use subversion to commit any approved changes and then check them out on the live boxes. We dont need to worry about shared file space and we get version control of the configuration as an additional benefit. Its similar for call recordings. We have call recordings going to a ram disc and then when they are complete there is a background process to copy them to the nfs volume. If nfs is unavailable then they are moved to the internal disk temporarily until the nfs is back online. We have never used this functionality but it add a little redundancy. I would put opensips at the front end which looks at the incoming destination number and routes the call to the appropiate front end asterisk box depending on the queue it should go to. The other asterisk box(s) will be a backup so if one asterisk box fails then one of the others takes over running that queue automatically. For call recording are you using mixmonitor? I would consider using the normal monitor command and pass these recordings off to the nfs and have another machine process them by mixing both legs and perhaps converting to mp3 aswell. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] Asterisk 1.8.15-cert3, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, 11.5.1 Now Available (Security Release)
On 27/08/13 19:20, Asterisk Development Team wrote: The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, and 11.5.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases http://www.asterisk.org/downloads/asterisk/all-asterisk-versions sais :- Latest Version - 11.2-cert2 However if you download it its the old cert1 version. Going direct to :- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ there is no cert2 version there yet. -- _ -- 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] How to get the original SIP result code
On 22/08/13 15:43, Mordechay Kaganer wrote:
B.H.
Hello, i'm using AMI Originate action (with async=true) to send
outgoing calls to a SIP trunk (using asterisk-java library to connect
to AMI).
The problem is that in case of failed originate, OriginateResponse
event is returning only the reason code which is sometimes not
sufficient to determine the real cause of failure. Also, there's no
way to link between the channel that was trying to dial and failed and
the original Originate request, because OriginateResponse is issued
only after the failed channel was hang up. Only successful
OriginateResponse will contain the unique id of the established channel.
Is there any way that my AMI application can get the original SIP
response of the failed Originate action?
I'm using Asterisk 1.8.22 and slightly tweaked asterisk-java
(https://blogs.reucon.com/asterisk-java/) 1.0.0.
You could dial a local channel instead and have that then go on and dial
the destination. You will then be able to retrieve the sip response
using something like :-
[localdial]
exten = _X.,1,Set(ddi=${CUT(EXTEN,,1)})
exten = _X.,n,Set(carrier=${CUT(EXTEN,,2)})
exten = _X.,n,Set(dialtime=${EPOCH})
exten = _X.,n,Set(_MASTERCHANNEL=${CHANNEL})
exten = _X.,n,Dial(SIP/${ddi}@${carrier})
exten =
_X.,n,Set(SIPcause=${MASTER_CHANNEL(HASH(SIP_CAUSE,${CDR(dstchannel)}))}, Responsetime=$[${EPOCH}-${dialtime}])
exten = _X.,n,Set(SIPcode=${CUT(SIPcause, ,2)})
However you will need to set storesipcause=yes in your sip.conf for
this to work. It is known to have a performance hit.
A better way would be to upgrade Asterisk and use hangup handlers. The
documentation on how to do this in Asterisk 11 is poor and often wrong.
I have written a blog about how we set it up as we use the feature all
the time.
http://gblades.blogspot.co.uk/2013/07/how-to-get-sip-response-code-in.html
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Re: [asterisk-users] How to get the original SIP result code
On 23/08/13 10:44, Mordechay Kaganer wrote: In my experience with local channels, they cause a huge performance problems, even without sipstorecause. We are dialing on about 100 channels in parallel and looks like this will kill my CPU :-) Really? We had Asterisk 1.8 with this setup and pushed 250+ concurrent calls through it and it coped quite well. All these used local channels for dialling. Our new systems are Asterisk 11 with no Dahdi. Quad Intel(R) Xeon(R) CPU E5-1410 0 @ 2.80GHz and reached about 40% cpu with 300+ concurrent calls. The new systems are a complete reinstall (OS upgrade with alter kernel to take advantage of res_timerfd) but like previous versions they still suffer from a gradual rise in load average. Every month or so we need to restart asterisk to bring the load average down. No idea what the cause is but its an automated process now so has no impact. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan MySQL inserted ID
On 20/08/13 17:48, Gergo Csibra wrote:
can I echo this variable ?
Like : exten = s,n,NoOp(${LAST_INSERT_ID()})
No, this is a mysql query, so:
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET C1=${ARG1},
C2=${ARG2}, timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
exten = s,n,MYSQL(Query resultid ${connid} SELECT LAST_INSERT_ID())
exten = s,n,NoOp(${resultid})
first is your original insert query, next you must read the
last_insert_id() mysql function with an other query, then you can
echo the resultid variable which contains the last inserted id.
I would be a bit concerned about doing this on a busy system. What would
happen if one call inserted a value, a second call inserted a value and
then the first call read the LAST_INSERT_ID? Would it get the wrong
value back?
If you do it in AGI then each query can have its own database connection
and so avoid this issue. If thats a problem use FastAGI and have a
daemon running and use transactions or another method to avoid the issue.
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Re: [asterisk-users] Cut off last character of EXTEN
On 20/08/13 09:29, Jonas Kellens wrote:
Hello,
how can I cut off the last character of the EXTEN-variable with
variating length ?
So I have :
112233#
123#
123456789#
I want to cut off the last character.
${EXTEN:-1} gives me #, but that is the character I want to cut off.
Set(variable=${EXTEN:0:$[LEN(${EXTEN})-1]})
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Re: [asterisk-users] Dialplan MySQL inserted ID
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
You could write an AGI script in something like php or perl and get it
to write to the mysql database instead. It can then set a variable which
the dialplan can pick up.
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Re: [asterisk-users] DAHDI wct4xxp high system CPU on idle?
On 14/08/13 18:48, Tony Mountifield wrote: I have a system running CentOS 5.9 and DAHDI 2.6.2 with a 2-port E1 card using the wct4xxp driver (also using Asterisk 11.5.0, but that isn't relevant to the question). With DAHDI and Asterisk started, the system appears to run normally, as far as I can tell from limited testing. I am monitoring User, System and Nice CPU usage using SNMP and MRTG, and I have noticed that when I have started up DAHDI, the System CPU jumps up to around 12% or so and stays there. It does this even if I don't start Asterisk. On previous systems I have built over the years, using CentOS4 and Zaptel, I don't recall seeing such high CPU usage just from having Zaptel started. It would be down near 0% until the system started handling real calls. So my first question would be: is this high CPU usage normal with current cards and DAHDI? It's curious that 12.5% is 1/8 of 100% and /proc/cpuinfo reports 8 CPUs, but I don't know whether that is just coincidence. The CPU is a X3450 with four cores and HT enabled. Any thoughts would be gratefully received! Cheers Tony One of our servers does that but only on one particular server. There is nothing special with the server as apart from being a different model Dell 1u due to being bought at different times its identical. It only has the one pcie expansion. We see the events/1 process taking 10% of system cpu. We use sangoma cards and this issue has persisted as we have upgraded from 1.4 to 1.6 and then 1.8 with Dahdi being upgraded at the same time. -- _ -- 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] App_meetme recordings
On 02/08/13 22:10, John Doe wrote: Can you create dynamic bridges like in meemet? Its the only way you can do it. Confbridge doesnt use a database or config file to store the conference details and pin codes. You need to manage all that yourself in the dialplan and then call confbridge to actually create or have callers added to a conference. More of a pain for very basic usage but much more flexible when hosting multiple conferences. -- _ -- 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] server for 500 concurrent SIP calls
On 05/08/13 12:38, Kamlesh Kumar wrote: Hi, Asterisk 1.6.2.9 PHP 5.3 Mysql 5.0 Can anyone suggest hardware specification for 500 hundred concurrent SIP only calls, no codec transcoding, no IVR, no Voicemail or so. Just plain switching. There is only one requirement is to execute one php script on call hangup (h extension) which will do some calculation and update the CDRs. We have a quad core 2.8Ghz machine which can handle 500 calls. It doesnt do codec translation but does a little voicemail and call recording plus a few other things such as day parting. AGI (fastagi) is run at call start, answer and hangup via perl daemons. -- _ -- 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] Dahdi interface flapping
The last two companies I have worked for have both had this problem with a BT ISDN30 line at some point. I also manage SS7 interconnects and its not unusual for there to be issues with them either. So dont assume its probably at your end :P On 02/08/13 15:12, Andre Goree wrote: I've contacted my telco for assistance but so far have been unable to come up with anything... -- _ -- 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] App_meetme recordings
On 02/08/13 17:10, John Doe wrote: Is there an easy way to have app_meetme create the recording in a temp location and move it once the conference is over? or should I just have a perl script run every minute to check for no users in the conference room and then move it? Asterisk 11 Thanks in advance! Why not use confbridge? the new version introduced in asterisk 10 runs in asterisk and so doesnt require DAHDI to work. This has performance benefits as a kernel timing source can be used. confbridge can also send out asterisk manager events so you can simple have something logged into the ami interface listening for just these events and then deal with the corresponding recordings as they come in. -- _ -- 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] Multi-homed SIP in Asterisk 11?
On 31/07/13 15:32, Tony Mountifield wrote: Most of my experience until recently has been in Asterisk 1.2, and I am just starting to make use of Asterisk 11 for new systems. I have a question about using SIP on a multi-homed machine. I have a customer who wants an Asterisk box with two network interfaces: one on the public Internet (no NAT), and one on a private LAN. The box will not do any IP forwarding between interfaces. They want to connect to a SIP trunk from an ITSP via the public interface, and to have SIP phones on their LAN registered via the private interface. I haven't tried such a setup before, so before creating a test system, I wondered if anyone here has made such a setup, and whether there are any issues with getting SDP contents and media routing correct? Cheers Tony I normally just ensure localnet= and externip= is set correctly. I normally also have 'directmedia=no' defined in sip.conf so that asterisk is performing store and forward for all the rtp traffic. That does mean rtp traffic for internal calls is going via asterisk where it could be direct between the phones but the amount of traffic doing this is normally pretty trivial so it doesnt matter in most cases. -- _ -- 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] Multi-homed SIP in Asterisk 11?
On 31/07/13 16:12, Tony Mountifield wrote: Thanks. But I thought localnet= and externip= were for when the external interface is going through NAT. In this case the ITSP is connected through a real non-NATted public interface. Is it possible to specify directmedia=no just for the SIP trunk? So that the phones could still do direct media between themselves, but not if they were connected to the trunk? Cheers Tony The localnet= can have implications as when asterisk sees that IP address it knows its local. This may have knock on effects when you have other settings which do different things of the endpoint is known to be behind nat. Its not clearly documented what needs to be set when directmedia=no is turned on. Does it only need to be set on one endpoint, both, or just the destination etc... I am sure you can do it that way but you will need to have a play and work it out. I wonder if using directmedia=nonat will help at all. It will disable directmedia if one endpoint is behind nat (this is where localnet= comes in) but I dont know if it will then know to enable direct media if both endpoints are behind nat. I suspect it wont as the endpoints may not necessarily be behind the same nat so this would be unsafe. -- _ -- 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] asterisk 1.4 freezes with queues and iax after virtualization
On 29/07/13 18:12, samuel wrote: there's no dahdi installed. Following debugging the issue, it looks like the astdb file is broken. Whenever database show command is executed it loops over the same results. The file itself is around 225K but dumping its content via asterisk -rx 'database show' creates and endless file. Is there any easy way to restore the database content? Thanks a lot for the replies, Samuel. There is some information listed here :- http://www.voip-info.org/wiki/view/Asterisk+database Are you actually storing any data in there yourself? If not it would probably be a lot easier to just rename the file and restart asterisk and it should create a new clean file. -- _ -- 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] using E1 PRI lines
On 30/07/13 05:35, Duncan Turnbull wrote: E1 PSTN line interfaces are either unbalanced 75 ohm( and used to use BNC connectors ) or a 120 ohm balanced twisted pair. The other standard is T1 and digium cards can let you choose between T1 E1 and definitely do 120 ohm Telco's will usually provide 120ohm twisted pair interfaces as it travels further and has less interference from noise. 120ohm is the standard for E1 RJ45 while T1 normally has a 100ohm impedance. The E1/T1 jumper on the Digium cards is actually changing the impedance. BNC is 75ohm and if your telco provides these you need a balun panel to convert the impedance and the wiring to RJ45. e.g http://www.tekmos.co.uk/baluns/panel/rj45/rj45-balun-panel-16e1-coax-front-mount.shtml -- _ -- 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] Dahdi interface flapping
On 30/07/13 15:36, Andre Goree wrote: /etc/dahdi/system.conf: span=1,0,0,ESF,B8ZS bchan=1-23 dchan=24 loadzone=us The first '0' in your span line above indicated that asterisk is generating the timing source. Normally the network operator provides timing so I would expect this to be a '1'. This can be the cause of the issue you are seeing as each end will be using a different clock which can be out of sync or drift causing data corruption, signalling errors and the d channel going down. -- _ -- 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] asterisk 1.4 freezes with queues and iax after virtualization
On 29/07/13 12:15, samuel wrote: Hi folks, Recently a customer of us moved his old asterisk installation, an 1.4.44 to a VMWARE infraestructure and has started having some weird issues. Asterisk started going slow and even refused to start up. After few tests, it only loaded when deactivating queues and iax2 (with noload in modules file). The thing is that it had been working with these modules loaded and lately it just freezes when trying to use these modules. We've made some checks to the server and there seems to be no issues with load, with swap, with wait (disk access), or other server parameters. Could it be some timing issues? How could we debug further the issue? Thanks a lot in advance, Samuel. Timing could be an issue. Is dahdi installed? Asterisk 1.4 is old and no longer supported. I would suggest upgrading which would also make the timerfd kernel timing source available if you are running on a recent operating system. See https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces -- _ -- 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] Asterisk CPU use
On 29/07/13 15:22, Eduardo Leones wrote: Hello, working in a call center where we set up a structure in asterisk. When my voip reaches 150 calls are with bad quality. We do not transcode codec. What I realized using the top command server (CentOS) processing is too high for the asterisk. But the general processor server is down. Would any limitation of Asterisk to use more hardware resources? tks Eduardo I think you need to press '1' in top so that it lists the cpu usage of each core. What version of asterisk are you running? What version of centos? Any dahdi cards installed? -- _ -- 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] using E1 PRI lines
On 29/07/13 16:28, Akib Sayyed wrote: Dear asterisk users I wanted to use E1 pri lines on my asterisk box but my provider support only 120ohm on E1 line. I dont know how to set those values. Please help me Its done on whatever interface cards you have. Some may have a jumper setting. I know Sangoma has it in their configuration file (wanpipe). -- _ -- 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] Sending 603 Declined message
On 26/07/13 16:32, Leandro Dardini wrote: In my dialplan I'd like to send a 603 Declined message to the user placing the call. I see the commands for the Busy and Congestion, but not the one for the Declined. Any help? Leandro I dont think you can. Normally you would use the Hangup() command with the hangupcause value that you wish to use. However there are no values which will be translated to a SIP/603. The closest would be Hangup(21) which would be 403 Forbidden -- _ -- 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] How to stop Dial from waiting for extra digits if number is incomplete.
We have Asterisk 1.6.2.17.2 connected to a sangoma E1 card. The problem
we are having is that we have a calling card type application and when
people enter the number to be dialled we call the Dial application. It
gets back an indication that the number is incomplete (via PRI cause
code 28 I assume) and waits for extra digits. If it gets extra digits it
appends them to the current extension and jumps to priority 1.
Is there any way this behaviour can be changed so if Dial tries dialling
an incomplete number it just fails and we can query ${HANGUPCAUSE} to
decide what to do.
Thanks
Gareth
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Re: [asterisk-users] New implementation asterisk
Edwin Blommaerts wrote: Hello everyone, I have some questions regarding implementing asterisk and my-sql. I’m no expert at asterisk but I’m going to list up my questions and hopefully someone will be able to help. -The people that setup our server made asterisk write the sound files onto the hard drive, and then somewhat later store these files into my-sql. Is this the proper way to do it? Or would it make more sense to just have my-sql store the path to the file? Or is it possible to store the data directly into my-sql from asterisk? So in general what is the best way to make this happen? Has anyone implemented something similar? Seems a little odd. We have voicemails stored on a separate server with access via nfs. Thats not to say you cannot store the files within the mysql database but it would make backups take much longer. With storing flat files you can just use rsync to do the backups so you actually only copy changed files which is much better. -They also claim that for performance and other issues no queries should be done on the main asterisk my-sql database, but instead we should setup replication between two my-sql databases and queries should be run on the slave. Is this needed and why is this needed, because my-sql prefers insert above read actions I don’t really see why? Our current server now shows segmentation errors at boot and they claim that this is due to queries on the database. I’ve also checked the query it returns the data to us in less then 3 seconds, so I’m doubting this can be the trouble. You do that so you can effectivly load balance between two boxes. One does all the inserts and database modification work while the others (you can have multiple slaves) handle all the reads. 3 seconds is a long time. I dont know what you are actually reading but if thats the time it takes asterisk to read a table entry then caller may be waiting 3 seconds before asterisk even attempts to dial the destination. -Sometimes the phone lines hang waiting, people can’t call out because asterisk seems to be holding on to these lines for a certain amount of time. What could be the cause of this? Do you do anything in the 'h' extension? If you are calling an AGI program there and it hangs that channel stays busy. -What is the best hardware architecture for asterisk implementation? My-sql and asterisk on separate servers, or doesn’t it really matter? It all depends on how many calls you are handling, how much voicemail access, how much custom agi scripts you run and various other things. If its database intensive then use a separate database server(s). If you are calling lots of AGI scripts then consider a separate fastAGI server and have the scripts running as a daemon so you dont have all the startup overhead. I really hope to get some advice, because, I’ve began to doubt/question the people who set the server up. I’m happy with any information, good references that you can give me. Thanks a lot in advance EB This email is scanned with Mcafee Groupshield. --- This email and any attached files are confidential and may be legally privileged. If you are not the intended recipient, any disclosure, reproduction, copying, distribution, or other dissemination or use of this communication is strictly prohibited. If you have received this transmission in error please notify the sender immediately and then delete this email. Email transmission cannot be guaranteed to be secure or error free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore is in no way liable for any errors or omissions in the content of this message, which may arise as a result of email transmission. If verification is required, please request a hard copy. -- _ -- 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] SIP DNS SRV
Jonas Kellens wrote: On 11/08/2010 09:50 PM, Jonas Kellens wrote: Hello, SIP DNS SRV records are not working. My Grandstream uses the SRV records to find the first Asterisk server to register to. This works. But when I shut down the Asterisk proces on server 1 and I restart my GXP 2010, the phone does not register to server 2... No mather how long I wait, there is no registration coming in... When I start the Asterisk proces again on server 1, then here registration comes in. Kind regards, Jonas. More info : [jo...@jonas ~]$ host -t srv _SIP._udp.sip10.domain.tld _SIP._udp.sip10.domain.tld has SRV record 25 10 5060 sip2.domain.tld. _SIP._udp.sip10.domain.tld has SRV record 5 10 5060 sip1.domain.tld. It sounds like the grandstream phones are not fully compliant with the SRV standard. They are probably just looking for the lowest priority entry and hardcoding that to be used all the time internally. If you restart the phone does it work? It might try the 25 priority entry if it cannot initially contact the primary server. -- _ -- 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] SIP DNS SRV
Jonas Kellens wrote: On 11/09/2010 02:12 PM, Gareth Blades wrote: Jonas Kellens wrote: On 11/08/2010 09:50 PM, Jonas Kellens wrote: Hello, SIP DNS SRV records are not working. My Grandstream uses the SRV records to find the first Asterisk server to register to. This works. But when I shut down the Asterisk proces on server 1 and I restart my GXP 2010, the phone does not register to server 2... No mather how long I wait, there is no registration coming in... When I start the Asterisk proces again on server 1, then here registration comes in. Kind regards, Jonas. More info : [jo...@jonas ~]$ host -t srv _SIP._udp.sip10.domain.tld _SIP._udp.sip10.domain.tld has SRV record 25 10 5060 sip2.domain.tld. _SIP._udp.sip10.domain.tld has SRV record 5 10 5060 sip1.domain.tld. It sounds like the grandstream phones are not fully compliant with the SRV standard. They are probably just looking for the lowest priority entry and hardcoding that to be used all the time internally. If you restart the phone does it work? It might try the 25 priority entry if it cannot initially contact the primary server. The way I test it : - Grandstream turned off. - Stop asterisk server1 (/sbin/service asterisk stop) - Turn on Grandstream (power up) Conclusion : Grandstream does not register. No register coming in on server2. Finally : - Start Asterisk again on server1 (/sbin/service asterisk start) Conclusion : Grandstream registers to server1. Jonas. Then it looks like the Grandstream phone dont fully support DNS SRV. maybe a firmware update will fix 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
Re: [asterisk-users] clustering
Use camailio or opensips as the registrar server so it accepts the sip registrations. You can have copies running on a couple of boxes using either a shared databases or a database on each server configured in master-master replication mode. Opensips can be configured to use the same database table that asterisk uses for authentication. Then you can use the load balancer module to send the call to whichever asterisk box has the most free lines. Normally you try and use opensips for most things such as call routing and registrations and leave asterisk to do the application type stuff such as conference calls and voicemail. Rizwan Hisham wrote: anymore ideas anyone please? On Fri, Oct 15, 2010 at 8:36 PM, Josef Grand josef.gra...@gmail.com mailto:josef.gra...@gmail.com wrote: use camailio for SIP SLB sip load balancer - Original Message - *From:* Rizwan Hisham mailto:rizwanhas...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Thursday, October 14, 2010 5:01 PM *Subject:* [asterisk-users] clustering Hi all, I am planning to do clustering for my company's asterisk servers. I dont know much about it, just read some articles on the internet and learned how to use DUNDi and some basic information about clustering. What I need to know is: 1. can i register end user with multiple asterisk servers at a time? 2. If not, Can I re-route registeration requests to different servers using 1 asterisk server as a gateway and multiple clustered asterisk servers behind it? cheers Thanks in advance -- Best Regards Rizwan Qureshi -- _ -- 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 __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a iti virifii par ESET NOD32 Antivirus. http://www.eset.com __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a été vérifié par ESET NOD32 Antivirus. http://www.eset.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 -- Best Regards Rizwan Qureshi -- _ -- 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] SIP DNS SRV
Yes that is the way it is supposed to work. You do have to rely on the sip devices you are using fully supporting SRV records though. Jonas Kellens wrote: Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is reachable again, so registration goes back to the production server ?? Do I need a low REGISTER timeout value for this to work ? Something like 60seconds, so the IP-phones register every 60 seconds... Kind regards, Jonas. -- _ -- 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] Asterisk to switch on electric heaters remotely?
Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only control 750W so you will probably need to get it to control a more powerfull relay as a heater is going to take a lot of current. It can be controlled by a virtual serial port so you just program the extension to make a system() call to a simple script which sends a string of characters to the serial port. That device is quite expensive. You could probably find something much cheaper on ebay. Gilles wrote: Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. 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] SIP DNS SRV
Yes if you set the production server to a higher priority then the backup then the RFC states that the client should periodically check and switch back to the primary server is it becomes reachable. Jonas Kellens wrote: Hello, I know YeaLink for example supports this... Can you tell me for sure that when the production Asterisk server becomes reachable again, the registration will go back to the production server ?? Jonas. On 10/18/2010 01:11 PM, Gareth Blades wrote: Yes that is the way it is supposed to work. You do have to rely on the sip devices you are using fully supporting SRV records though. -- _ -- 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] Create channel bank with TDMoE
Karim Davoodi wrote: Hello, I want to create channel bank in this case: channel bank |-| | FXS,FXO-TDMoE--|--Asterisk |-| How can it? http://www.voipon.co.uk/redfone-fonebridge2-quad-t1e1-ethernet-bridge-p-348.html -- _ -- 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] MYSQL ADDON INSTALLATION ERROR
Rizwan Hisham wrote: Hi All, Please refresh my memory. I am trying to install asterisk after 2 years. I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 1.8.0-rc2 on centos 5.5 but getting the following errors. app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory app_mysql.c: In function ‘mysql_ds_destroy’: app_mysql.c:135: warning: implicit declaration of function ‘mysql_close’ app_mysql.c:138: warning: implicit declaration of function ‘mysql_free_result’ app_mysql.c: In function ‘aMYSQL_connect’: app_mysql.c:319: error: ‘MYSQL’ undeclared (first use in this function) app_mysql.c:319: error: (Each undeclared identifier is reported only once app_mysql.c:319: error: for each function it appears in.) app_mysql.c:319: error: ‘mysql’ undeclared (first use in this function) I think i have seen these errors before and did manage to get rid of them but I cant remember how i did it and even dont remember the reason for these errors. Looks like a header file for mysql addon is missing which is actually missing (i have checked). How am I suppose to find it? Plz help. -- Best Regards Rizwan Qureshi Make sure you have the mysql-devel package installed. -- _ -- 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] How to test BIG traffic through DAHDI/WANPIPEinterfaces
As the previous poster said use the sip software to make test calls. Have the number it dials go out of the sangoma card and back into another port via a crossover cable to an extension which answers and plays back a file for a second or so before hanging up. You can then make lots of calls which constantly make outgoing calls on 4 ports and incoming calls on another 4 ports. By being able to change the diration of the call to can load the box very well. Danny Dias wrote: ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl mailto:i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- 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
