Re: [asterisk-users] Upgraded server crashes on voicemail storage
On 06/09/2017 12:37 PM, Mike Diehl wrote: Well, I guess my assumption has been proven wrong. It is NOT the odbc drive. I recompiled Asterisk w/o odbc voicemail storage and I'm still getting crashes when someone leave voicemail. This is probably not it BUT. A long time ago, voicemail lost it's mind when codecs were changed and they did not exist in the config file. Maybe the config file was changed during the upgrade? Maybe test it with a fresh voicemail db? Adrian Serafini -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
Maybe shut off qualify for the peer? I think I tried twinkle a few years ago and it didna (yes didna) like the qualify packet. the sip options qualify packet is only needed to keep the UDP state tables in a firewall if the peer is remote -- _ -- 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] Load Balancing with DNS SRV without DUNDI
On 05/24/2015 11:01 PM, Mehdi Shirazi wrote: Hi I want to load balance SIP calls between two(or more) Asterisks with only DNS SRV. I used bidirectional sync Unison to synchronize configuration files and internal database file between two Asterisk boxes. The problem is when a calls come to Asterisk1 but SIP endpoint is registered on Asterisk2.How we can check a SIP endpoint is registered or not and what is Contact information in Dialplan ? Regards babak If you used Opensips with a Mysql backend. The two Opensips servers could query a command db with the contact URI. -- _ -- 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 trunk no audio
But the phone rings - so its routed - just no audio. The ringing is SIP signaling. The audio is RTP data. See if the audio is getting routed with a sniffer. Maybe use one codec that both clients support. Adrian Serafini -- _ -- 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] SMS Capabilities
Just amazing! A thing of beauty yeah? We will be going with the following card: http://www.voipon.co.uk/openvox-g400p4-p-1150.html Hope setup and configuration will go easy? Jayson You send/receive sms directly to a VoIP provider. The provider has to support it. I sent and received to/from polycom and cell phone. I didn't test: mms, a Q'ing mechanism if the cell/VoIP phone were unavailable. I used the native sms app, it never failed on about 20 tests. Adrian Serafini -- _ -- 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] ast_writefile: No such format 'h261', yet h261 is the only video format that works.
If h261 is checked in ekiga's video format list I have video, and [Mar 21 16:25:32] WARNING[31818][C-0010]: file.c:1241 ast_writefile: No such format 'h261' Ekiga can do SIP. Maybe try that? And set/prioritize the codec in ekiga to desired codec, not h261. -- _ -- 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] Need more meetme users -- hitting some limit
Coincidentally, 512 is my target. Any clues on how to get 200 more? Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2 luddite? I'm a big fan of older releases with 1 year plus of uptime. -- _ -- 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] ast_writefile: No such format 'h261', yet h261 is the only video format that works.
On 03/21/2014 02:09 PM, David Woodfall wrote: H.323 is a communications protocol like SIP. H261 is a codec like ulaw or gsm. You do not need H323 unless you are using the H323 protocol INSTEAD of SIP. I see. In Ekiga video codec window they are listed like: [ ] h26190kHz H.323. SIP Ok so your all SIP. Find the command to show the codecs for your release. The wiki has info to point you in the right direction. For old 1.4 releases, I set the codec in the sip.conf file peer. Also try another SIP video phone maybe on android? -- _ -- 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] modify from field sip headers
Im trying to modify the 'From' field in my sip headers in order to
include extra info (user=tel) as it follows:
The default extensions.conf has this, it might help.
;---
; from-pstn-to-did
;
; The context is designed for providers who send the DID in the TO: SIP
header
; only. The format of this header is:
;
; To: sip:2125551212@172.31.74.25
;
; So the DID must be extracted between the sip: and the @, which this does
;
[from-pstn-toheader]
exten = _.,1,Goto(from-pstn,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
;---
--
_
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Re: [asterisk-users] what is actually a trunk in a sip trunk?
On 03/10/2014 07:39 PM, Thomas Rechberger wrote: no trunking or bonding involved, so why just everybody calls this a trunk? It is just another SIP peer. You tend to route more than one extension down/from 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] how to provision asterisk's phonebook to Polycom vVX310's
On 01/21/2014 01:55 PM, Stanley van Dijk wrote: Hi, Am running a freepbx install and created trunks, extensions and groups. Now I'd like to hand out the Asterisk phonebook to the phones (all VVX 310's). Is there an easy way to do this? Best, Stanley Even the old ones could view a webpage. Have a script read the Mysql DB/users table data, then output in XML. The newer ones can output in HTML5. This solution is auto updated when you add GUI users. Or you could maintain a static directory, but this is not good for a large office. The maintenance is impossible. Polycom used to charge for LDAP directory access, this might be free now? Or maybe I dreamt about free LDAP while reading a release note. :) Adrian -- _ -- 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 read IRQs and timing slips values
On 01/14/2014 04:32 AM, Olivier wrote: I'm 100% sure my PBX is configured to use provider's clock (but I won't swear my PBX is currently using provider's clock) I have had to power the server down, UNPLUG the power, leave unplugged for 4 minutes, power up. I had a T1 timing issue this procedure fixed. Adrian -- _ -- 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 read IRQs and timing slips values
On 01/13/2014 11:39 AM, Shaun Ruffell wrote: If you have another board, yes, you could try. But I would recommend checking all your cables, etc. Also, while highly unlikely, I've heard of cases in the past where some smaller providers were expecting to source timing from customer premise PBX (since they were acting as a SIP gateway on the backend). Check the T1 cable doesn't pass any high EMI area's like a power supply. Adrian -- _ -- 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] *8 and SIP
On 12/31/2013 12:41 PM, Vladimir Mikhelson wrote: Nick, You may want to try *97 and *98 to access voice mail. Regards, Vladimir On 12/31/2013 10:23 AM, Nick Olsen wrote: Greetings all, First time poster, Sorry if this has been answered here before. We recently replaced a failed 1.4x asterisk PBX at a customer location. Voicemail access was setup when the customer dialed *8, This worked in 1.4. Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon). The strange part is *8 no longer works. The only CLI feedback I get is == Using SIP RTP CoS mark 5 In features.conf, Callpickup *8 is commented out, But just incase I also changed it to *7 (We don't use that feature). It appears to be something completely SIP based, As if the call originates from DAHDI, It works fine.. Maybe it's a context issue. Check the dialplan context for the *8 logic. Crank up the verbosity of the CLI and make a test call. You might have to reboot after the features.conf change. Adrian -- _ -- 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] iax2: no authentication, but still peer?
The qualify is on for the peer. It is failing to reply to the requested SIP status. Maybe it is on wifi, screen goes off, wifi follows, zoiper iax stack doesn't re-reg with the asterisk. [Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper: Peer 'n4' is now REACHABLE! Time: 441 [Oct 8 18:15:58] NOTICE[519]: chan_iax2.c:8153 register_verify: Host n4ipaddr failed MD5 authentication for 'n4' (c374d0a70c72e6e9bd359aa6a0f1a6c2 != 2c76c104bbfc3d54f566490f40cd12bd) [Oct 8 18:19:17] NOTICE[517]: chan_iax2.c:11077 socket_process_helper: Peer 'n4' is now TOO LAGGED (1002 ms)! [Oct 8 18:19:29] NOTICE[512]: chan_iax2.c:11071 socket_process_helper: Peer 'n4' is now REACHABLE! Time: 300 [Oct 8 18:26:02] NOTICE[519]: chan_iax2.c:11077 socket_process_helper: Peer 'n4' is now TOO LAGGED (1017 ms)! ip-172-31-29-115*CLI iax2 show peers Name/Username Host Mask Port Status Description n4 n4ipaddr (D) 255.255.255.255 4569 LAGGED (1017 ms) is it still registered, or do we really have an authentication problem? sean -- _ -- 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 sqlite3 for CDR logging
faster than using MySQL. Has anyone ever benchmarked this to quantify Put Mysql on another machine and network the db service. -- _ -- 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] Transfer Fraud
On 09/13/2013 04:12 PM, jg wrote: Is there a general recipe to avoid fraudulent calls under the following conditions? A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to call and then transfer to him), i.e. the Dial cmd for the internal context contains Tt. Then an outside call would operate as a Local channel in an internal context after the first transfer. If the internal context allows to dial outside, which is quite common, then this can be abused by the outside caller. An obvious solution is to disallow Local channels to call outside lines, but there are some possible side effects if Local channels are used explicitly. This would require adding a persistent channel variable (the ones with __). create a separate context for outbound calls. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] crossed channels
Exactly, mixed audio, callers are linked to the call of another caller,the calls are interlaced, is something that happens sometimes... It can happen with analog dahdi calls. If this is the case, start inbound on one end of the group, outbound from the other end. Adrian -- _ -- 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] Japanese voicefiles
Hi Chris, Thanks for replying, I've got it set in the context in extensions.conf: [TokyoReception] exten = s,1(TOKYORECEPTION),Answer exten = s,n,Set(CHANNEL(language)=jp) ; set japanese by default exten = s,n,SET(LOOP=0) exten = s,n,SET(LANG=JP) It could be something fixed between 1.4.18 and 1.4.21. Wish I could find the bug ID now... Can you confirm you set the language the same way ? If you've got files in ..sounds/britishfemale, then how are you setting the sub-folder ? (I thought it would only choose en, fr, jp, etc based on country codes). If I put a custom vm-dialout.sln file in sounds/jp, then it does play that file, so it seems to only affect the sounds/jp/digits folder (a sub-sub folder with numbers). Thanks, Adrian -- _ -- 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] Japanese voicefiles
Ok... I'm baffled.. I took a copy of my machine and put it in a virtual machine, then upgraded the VM to 1.4.44 to experiment, and unknowingly let it install the default US GSM sounds again. My code runs, but, it still plays the US digits when the debug says the below. You can see its set to JP, and that its picking normal voiceprompts from JP I tried deleting the 2.* files from sounds to force it to error (and confirm which file its playing), but it doesn't error. SAYDIGITS just skips the numbers it can't find :| [2012-08-24 11:33:31] VERBOSE[18633] logger.c: -- Executing [9222@TokyoReception:1] SayDigits(SIP/XXX-005, 9222) in new stack [2012-08-24 11:33:31] VERBOSE[18633] logger.c: -- SIP/XXX-0005 Playing 'digits/9' (language 'jp') [2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing [9222@TokyoReception:2] NoOp(SIP/XXX-0005, LoopCounter is 0) in new stack [2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing [9222@TokyoReception:3] GotoIf(SIP/XXX-0005, 0?dialit) in new stack [2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing [9222@TokyoReception:4] GotoIf(SIP/XXX-0005, 0?dialit) in new stack [2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing [9222@TokyoReception:5] GotoIf(SIP/XXX-0005, 0?dialit) in new stack [2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing [9222@TokyoReception:6] GotoIf(SIP/XXX-0005, 0?dialit) in new stack [2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing [9222@TokyoReception:7] GotoIf(SIP/XXX-0005, 1?dialit) in new stack [2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Goto (TokyoReception,9222,13) [2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing [9222@TokyoReception:13] Playback(SIP/XXX-0005, vm-dialout) in new stack [2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- SIP/XXX-0005 Playing 'vm-dialout' (language 'jp') [2012-08-24 11:33:34] VERBOSE[18633] logger.c: == Spawn extension (TokyoReception, 9222, 13) exited non-zero on 'SIP/XXX-0005' Thanks, Adrian -Original Message- From: Adrian Marsh Sent: 24 August 2012 09:42 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Japanese voicefiles Hi Chris, Thanks for replying, I've got it set in the context in extensions.conf: [TokyoReception] exten = s,1(TOKYORECEPTION),Answer exten = s,n,Set(CHANNEL(language)=jp) ; set japanese by default exten = s,n,SET(LOOP=0) exten = s,n,SET(LANG=JP) It could be something fixed between 1.4.18 and 1.4.21. Wish I could find the bug ID now... Can you confirm you set the language the same way ? If you've got files in ..sounds/britishfemale, then how are you setting the sub-folder ? (I thought it would only choose en, fr, jp, etc based on country codes). If I put a custom vm-dialout.sln file in sounds/jp, then it does play that file, so it seems to only affect the sounds/jp/digits folder (a sub-sub folder with numbers). Thanks, Adrian -- _ -- 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] Japanese voicefiles
Ok, This is something to do with folder layouts. I have: /var/lib/asterisk/sounds - uk files /var/lib/asterisk/sounds/digits -uk/us digits /var/lib/asterisk/sounds/jp - Japanese files /var/lib/asterisk/sounds/jp/digits - Japanese digits I read the 1.4 notes on : http://www.voip-info.org/wiki/view/Asterisk+multi-language Which says that, in 1.4, by default it'll work as 1.2, which expects: /var/lib/asterisk/sounds - uk files /var/lib/asterisk/sounds/digits -uk/us digits /var/lib/asterisk/sounds/digits/jp - japanese digits But if you put languageprefix=yes in asterisk.conf (and restart I presume), then it would work the 1.4/1.6 way But I've now got languageprefix=yes set in my test setup, and I can only get Japanese digits if I put them in sounds/digits/jp. Odd, but at least I've a solution, sort of... Will just test now in my old 1.4.18 setup with a symbolic link I think -- _ -- 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] Log faulty calls?
I ended up writing a basic parsing script that lets me search the full log, based on some unique identifier (eg, my own extension vlog 2027). It then digs out the associated A*k log number for each line that's it, and lists them out. Then I choose the 'call' and it re-filters by that call only. Its not perfect, as asterisk rolls log numbers over, but works well enough if I want to dig out just the logs for one call. Its not automated in any way though, I just use it for manual debugging. Thanks, Adrian From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 24 August 2012 14:17 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Log faulty calls? Actually, you could look for WARNING or ERROR and probably find what you needed. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF Sent: Friday, August 24, 2012 8:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Log faulty calls? Thank you Danny, but the problem is that I don't know what exactly I shall look for. I think there's no specific word in the log that clearly identifies this kind of problem? ): 2012/8/24 Danny Nicholas da...@debsinc.commailto:da...@debsinc.com Not the best solution, but you could do a quick and dirty crawler to query /var/log/asterisk/full in PHP or PERL or your language of choice. Even in a 4K-5K calls per day environment this process usually takes less than 1 minute to run. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF Sent: Friday, August 24, 2012 7:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Log faulty calls? If somebody is calling me using a wrong configured SIP phone, he gets back an error message from my Asterisk server. That's ok, however I'd also like to know that I missed a call. However there's no CDR entry created in that case and checking the asterisk logs manually is not that great... Any way to get CDR records (or any other way of noticing it) even if a call gets declined through to a wrong configured sip phone? Thanks and best regards Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Japanese voicefiles
Hi Guys, I've a few questions around languages I'm on 1.4.18 (old yes I know, but upgradings not an option just yet). I've downloaded the gsm Japanese files from ftp://ftp.voip-info.jp/asterisk/sounds/ and put them in place I've found that when I switch to jp, and play some of my own voicefiles in Background and Playback, that it chooses the /var/lib/asterisk/sounds/jp folder files and plays them, but, voicemail doesn't seem to do this, instead it picks the English files (although the debug output says its using 'jp'). I've seen references to a patch for this, but any idea where the patch is ? Secondly, I'm trying to open .gsm files in Audacity (in particular these japanese ones, so I can confirm they are Japanese), but I just can't get the audio format right (audacity 2) Open RAW: Encoding ? Byte Order ? Channels: mono, Sample rate 8000hz. I've set my Pref Quality defaults to 8000hz and 16-bit, but I think that's only for recording. Anyone know the correct setting? I've been able to play them in Quicktime so I think they're ok, I just want to see them in Audacity. Thanks, Adrian -- _ -- 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] twenty thousands (20, 000) users, which asterisk and how many servers?
AsteriskNOW is a GUI on top of Asterisk; it does not change the ability of the system to handle call load. I thought the AsteriskNOW GUI was now a FreePBX clone. If so, every call now uses a perl script to make the call. This is considerably more overhead than a dial-plan written in native asterisk code. For the 20,000 calls, I would use Opensips for the SIP and Asterisk for audio playback, transcoding, voicemail, fun. Adrian -- _ -- 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 create a module
Thanks Steve!... I will try the tip and I will let you know the result. Adrian Abramovich On Fri, Jul 8, 2011 at 12:26 PM, Steve Murphy m...@parsetree.com wrote: On Fri, Jul 8, 2011 at 10:39 AM, Adrian Abramovich adrianabramov...@gmail.com wrote: Hi, We are using asterisk 1.4 and we use a Perl script to record some specific calls. As far, everything is working well. I was thinking about create a module in order to improve script's performance. I checked the Russell's blog: http://www.russellbryant.net/blog/2008/06/19/how-to-write-an-asterisk-module-part-1/ This is a old post and I would like to know if there are something new. What I do, is look at the other apps/funcs for guidance. Pick the smallest first, and you can copy their style and layout. The module spec has evolved from 1.4 to 1.6 to 1.8, but it's the same basics (to a degree). Is it a good idea to move to module? If it increases performance, and you need that, then heck yes! The only drawback is that it *is* in the source; you'll have to tweak it as you move up the versions. You have to compile and install it. Perl would remain static (I would imagine) even when you update asterisk. Thanks in advance, Adrian Abramovich -- _ -- 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 -- Steve Murphy ParseTree Corporation 57 Lane 17 Cody, WY 82414 ✉ m...@parsetree.com ☎ 307-899-5535 -- _ -- 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] HA Asterisk
On 04/30/2011 10:20 AM, Adolphe Cher-aime wrote: You can't do PRI failover while using internal PRI cards. To do so you need a standalone PRI box a good one i use often is foneBridge from Redfone. U can use foneBridge as follow Hi, You can do a PRI failover with Dataprobe switches. Use the monit daemon to check for red alarms in syslog, then shutdown asterisk, then shutdown the PRI, the backup PRI is auto switched through the Dataprobe. Adrian Serafini -- _ -- 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] One PRI card with 2 (or more) Telcos
Is there a problem having 2 telcos on the same PRI card? I think you go with one master timer as the Telco. Then the other spans are secondary, tertiary, quaternary timers. Adrian -- _ -- 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] OT: OpenSIPS vs Kamailio -- which do you use and why?
Hi, We use Opensips and like the results. The forks are similar, docs from one can help in the other. The opensips mailing list is monitored by one of the main developers. He is even in the IRC chat in the mornings. The docs are kept current on the opensips webpage. They like to change modules a bit, so really watch your versions. The commercial PDF Building Telephony Systems with OpenSIPS 1.6 is excellent.(duck) Yum is nice for the dependencies, but I would use a compile for Opensips. Most of the docs are Debian specific. I love Debian, but our clients love Centos. I have some Centos Opensips compile docs if needed. There are a few GUI's, but I prefer Opensips-cp. To put opensips-cp on a remote server, you need the xmlrpc module loaded on opensips. This works in Debian but fails on Centos (64 bit ONLY). Good luck, Adrian On 03/04/2011 01:49 PM, Steve Edwards wrote: I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? -- _ -- 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] Panasonic KX-TGP500 w/Asterisk
Has anybody have any experience using the KX-TGP500 Cordless DEC 6.0 System with asterisk ? Hi, They sent us a few for free, so I guess I owe them a review. The Panasonic 500 and 550 worked very well with *sterisk 1.4. The full 550 phone seemed a little small, but all the advertised features worked. Audio quality was very good with a nice long range. The cordless phones are light weight with a belt clip, You just register the base set or phone, then connect the cordless phones. It pulls a config from central server, firmware options are thorough for business needs. One cordless phone (out of 5) developed a bad key. I really have to press hard to get the 1 key to work. The key went bad after 5 months. Overall, I really like the phones. Adrian Serafini -- _ -- 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] Meetme and MOH
Thanks all, I realised after posting 2 things.. 1) I needed to also cover MOH outside of meetme. And that 2) theres a bug in 1.4.18 where the defaults aren't reloaded properly for MOH, and you have to do a server stop/start to get them to reload. Thanks, Adrian -- _ -- 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] Meetme and MOH
Yes John... but I also now find in testing many things broken between my IAX provider and 1.4.37 Which is a reason to hold back... Thanks, Adrian From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: 26 November 2010 13:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Adrian Marsh Subject: Re: [asterisk-users] Meetme and MOH Adrian Marsh wrote: Thanks all, I realised after posting 2 things.. 1) I needed to also cover MOH outside of meetme. And that 2) theres a bug in 1.4.18 where the defaults aren't reloaded properly for MOH, and you have to do a server stop/start to get them to reload. Thanks, Adrian Probably why there is a 1.4.37? I found many things broken between 1.4.13 and 1.4.21 But that is now ancient history John Novack -- Dog is my Co-pilot -- _ -- 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] IAX inbound failing
Hi, I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it into production. Ive done this by installing 1.4.18 onto the VM, putting my config files in place and then installing 1.4.37 over the top (which is what I'd have to do on production). I've found a few issues in the config files, but nothing I couldn't handle until... I hit inbound IAX issues. My Voip provider has already told me that they don't use tokens, so I've added requirecalltoken=no into iax.conf However when a call is placed and routed to the VM, nothing appears in the CLI to show the call coming in. If I turn on IAX debug, I can see an initial message, and if I take that requirecalltoken line out and remake the call, then I get an error from Asterisk telling me that I need the line in to process the call - so the call is getting to Asterisk (not a firewall issue then). The call logic was working fine prior to the upgrade, and as I don't get one line of output at the CLI I'm wondering what is going on! I can see the INVAL in the debug, but I'm not sure what that's actually meaning - is it authentication?? The username/secret are all valid. My iax.conf entry looks like: [inboundcontext] type=user context=incomming_pstn username=inboundcontext secret=xx host=dynamic trunk=yes requirecalltoken=no Any ideas ? Thanks, Adrian IAX Trace: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 7ms SCall: 6 DCall: 0 [212.11.91.201:4569] VERSION : 2 CALLED NUMBER : 2095 CODEC_PREFS : () CALLING NUMBER : 01793xx CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: LANGUAGE: en USERNAME: inboundcontext FORMAT : 8 CAPABILITY : 65407 ADSICPE : 2 DATE TIME : 2010-11-25 17:01:46 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00011ms SCall: 00403 DCall: 6 [212.11.91.201:4569] AUTHMETHODS : 3 CHALLENGE : 167512360 USERNAME: inboundcontext Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 00403 [212.11.91.201:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 7ms SCall: 6 DCall: 0 [212.11.91.201:4569] VERSION : 2 CALLED NUMBER : 2095 CODEC_PREFS : () CALLING NUMBER : 01793xx CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: LANGUAGE: en USERNAME: inboundcontext FORMAT : 8 CAPABILITY : 65407 ADSICPE : 2 DATE TIME : 2010-11-25 17:01:46 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 7ms SCall: 09750 DCall: 6 [212.11.91.201:4569] AUTHMETHODS : 3 CHALLENGE : 151094941 USERNAME: inboundcontext Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 09750 [212.11.91.201:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 09835ms SCall: 6 DCall: 0 [212.11.91.201:4569] Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 09835ms SCall: 6 DCall: 0 [212.11.91.201:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 7ms SCall: 6 DCall: 0 [212.11.91.201:4569] VERSION : 2 CALLED NUMBER : 2095 CODEC_PREFS : () CALLING NUMBER : 01793xx CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: LANGUAGE: en USERNAME: inboundcontext FORMAT : 8 CAPABILITY : 65407 ADSICPE : 2 DATE TIME : 2010-11-25 17:01:46 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00015ms SCall: 10434 DCall: 6 [212.11.91.201:4569] AUTHMETHODS : 3 CHALLENGE : 102034710 USERNAME: inboundcontext Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00017ms SCall: 02331 DCall: 0 [82.71.203.26:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 10434 [212.11.91.201:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00017ms SCall: 02331 DCall: 0 [82.71.203.26:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 19834ms SCall: 6 DCall: 0 [212.11.91.201:4569] Rx-Frame Retry[ No] -- OSeqno: 003
Re: [asterisk-users] Someone has hacked into our system
Hi Gary, I went through this process a few times over the past few years. Theres a few short guides for securing Asterisk, but much of it depends on your design. If it's a traditional POTs-type PBX then locking down IPs using firewalls is a great thing, however if you make use of inbound-SIP calls from end-user PC clients on the Internet then that's not always possible. So heres my recommendations: 1) Change the default context name to something like publicinbound. 2) Create a context called publicinbound that does basically nothing. 3) Setup a different context for an peer or friend IAX or SIP, or whatever. That way you can see which connection the hackers coming in from. 4) If you don't want to firewall off the whole internet, then at least make use of fail2ban - it's a free scripted addon that watches for hacking attempts and firewalls them off. 5) Really really long passwords and usernames - this ones pretty key. My first task was in going through and understanding where all the passwords were and changing them. I now make mine completely random and a min of 30 chars. 6) IP restrictions. If a peer or user does have a fixed IP, then define it in the appropriate config file. 7) The alwaysauthreject is good.. helps fumble the hackers. Thanks, Adrian -- _ -- 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] Meetme and MOH
Hi, With a dynamic Meetme using: MeetMe(|DsMrc) How do I control which context MOH uses, other than default ? Asterisk: 1.4.15 Thanks, Adrian -- _ -- 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] friend, peer confusion in sip.conf
Hi, I'm trying to create a link between two PBXs. One is Asterisk 1.4.15, the other is an unknown 3rd party PBX. In my internal testing, beween two A*k servers, I found that if I created two sip accounts from the same IP, one as peer and one as user (intending to give an -IN and -OUT setup), then inbound calls always seemed to route via the -OUT account and failed. My fix was to use type=friend, which seemed to make sense and be ok. Now with the 3rd party PBX, if I set type=friend, then we get an error of Peer is not supposed to register. If I then set type=peer, it registers ok... But I thought that friend=peer+user ?? Thanks, Adrian -- _ -- 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] Context issue
Hi,
Running 1.4.15. I've a SIP user as below. My default context in
sip.conf is [incomming_pstn]
I'm having trouble with inbound calls going to the wrong context.
[test-ubi]
username=test-ubi
type=friend
secret=XXX
host=dynamic
canreinvite=no
context=testinbound
nat=yes
allow=ulaw
allow=gsm
allow=alaw
qualify=no
the testinbound context includes the code to prepend a 2 to the CLI
before passing it onto another context
[testinbound]
exten =
_,1,ExecIF($[${RECORDSIP}=TRUE],Monitor,wav|${TIMESTAMP}-${CALLE
RID(num)}-${EXTEN}-${UNIQUEID}.WAV)
exten = _,n,NoOp(REWRITE CALLERID)
exten = _,n,ExecIf($[ ${LEN(${CALLERID(num)})} = 4
]|Set|CALLERID(num)=2${CALLERID(num)})
exten = _,n,Goto(local,${EXTEN},1)
However, when a call comes in, its being passed to the
[incomming_pstn] context instead of [testinbound].
The Outbound server is dialling:
-- Executing [114...@from-sip-uk:2]
Dial(SIP/235012071833427-0a068a18, SIP/test-ubi/4201|40|r) in new
stack
-- Called test-ubi/4201
And that test-ubi account on there has the same SIP account setup.
The inbound server seems to skip the testinbound context completely
though, jumping straight to incomming_pstn, but I've no idea why.
I think it should be going to the context defined in test-ubi
ubiphone*CLI
-- Executing [4...@incomming_pstn:1]
Answer(SIP/192.168.50.132-b7d4f6b0, ) in new stack
-- Executing [4...@incomming_pstn:2]
SayDigits(SIP/192.168.50.132-b7d4f6b0, 2333) in new stack
-- SIP/192.168.50.132-b7d4f6b0 Playing 'digits/2' (language 'en')
-- SIP/192.168.50.132-b7d4f6b0 Playing 'digits/3' (language 'en')
.
But any idea why ???
Thanks,
Adrian
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Re: [asterisk-users] Context issue
How odd... If I specify the host=dynamic then it goes to the wrong context. If I specify the host=192.168.50.132, then it goes to the correct context. If I don't specify the host at all, then it also goes to the correct context... (but then of course I can't use that account for outbound calls..) Adrian -- _ -- 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] 3xx redirect response list Noop and capture
Hi, I Just setup asterisk to send SIP calls to a SIP redirect server that response back with a list of destinations, if the first destination is not able to terminate the call, asterisk does not try the second , it just hang up. How can I Noop and capture the list inside the 3xx response?, for storing it and then by using Dial status, I will be able of failover through that list and hang-up only after trying the whole list. This is the approach I figured, but I gladly accept any other suggestion. Thanks in advance. Adrian -- _ -- 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] Blind transfer feature
Hi, Am running 1.4.18 at the moment, and am trying to implement inline blind transfer. I have : [featuremap] blindxfer = *6 ; Blind transfer in features.conf And in extensions .conf under [globals] : DYNAMIC_FEATURES=automon#blindxfr So what am I missing ?? Have read through http://www.voip-info.org/wiki/view/Asterisk+config+features.conf Thanks, Adrian -- _ -- 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] AMR codec for Asterisk 1.6.1.X
It says in the readme from that link you provided: This patch adds AMR-NB support to Asterisk 1.4 (for Asterisk 1.6 check out asterisk 1.6 branch and use the asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov)) Did you use the 1.6 branch and patch ?? I'll have to try this myself at some point. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrea Cristofanini Sent: 05 May 2010 14:22 To: asterisk-users@lists.digium.com Subject: [asterisk-users] AMR codec for Asterisk 1.6.1.X Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - -- - - - - - - - - - gsm - - 2 22 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 12 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 -2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2- 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 22 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 11 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - -- - - - - - - - - - ilbc - - - -- - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 1200112001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats version debbi*CLI core show co codec codecs config debbi*CLI core show code codecs codec debbi*CLI core show codec codecs codec debbi*CLI core show codec audio Usage: core show codec number Displays codec mapping debbi*CLI core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) debbi*CLI The CLI does not show codec audio or codedc translation for AMR NB. Anyone have any idea ?? Thanks in advantage Andrea -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP Termination in Japan
Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -- _ -- 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] Repeated: Got SIP response 489 Bad eventback from
Hi James, Thanks for the help. 3.10 registers into my SIP server just as a normal SIP client. Yes, qualify=yes. I just tried setting that to no on my end, and I still get the message. I'll try turning it off on 3.10 too tomorrow and capture some trace too Adrian Hi All, I've two asterisk servers on the same LAN, both 1.4, and I keep getting Got SIP response 489 Bad event back from 192.168.3.10 No idea whats causing it. The only references I can find mentions NATing issues, but these are on the same LAN so NAT shouldn't be an issue. 3.10 does authenticate into the server logging the error. The error appears in the log every 1m20s (ish) Is 3.10 on a SIP trunk to the other asterisk box? Is qualify=yes on this SIP trunk? I think you'll find that if you run an ngrep/tcpdump on port 5060 on the box receiving the error it will send out an OPTIONS or NOTIFY (I can't remember which) and then you'll see the 489 Bad Event. Grab a trace of the SIP traffic and post it, its the only way to know for sure though. -- James Any ideas? Thanks, Adrian -- _ -- 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 -- _ -- 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: Got SIP response 489 Bad event back from
Hi All, I've two asterisk servers on the same LAN, both 1.4, and I keep getting Got SIP response 489 Bad event back from 192.168.3.10 No idea whats causing it. The only references I can find mentions NATing issues, but these are on the same LAN so NAT shouldn't be an issue. 3.10 does authenticate into the server logging the error. The error appears in the log every 1m20s (ish) Any ideas? Thanks, Adrian -- _ -- 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] (no subject)
Hello, I'm looking for some advice on securing Asterisk. Recently my servers been under several brute-force SIP attacks. I have several remote sites, as well as many roaming users, who may have PC softclients and/or SIP based hardphones. My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone quite widely open. Does Asterisk 1.6 have anything in it that can automatically block out an attacking IP, say if it receives several 20 or so failed attempts from that IP in x minutes? I haven't looked at Secure SIP in quite a while, is that now integrated into 1.6 ? One thing that's confusing me in my config, is that I thought that if I set NAT=no in sip.conf, then I wouldn't be able to connect to that SIP account unless I was on the local LAN, specified by locallan= However in some testing, I'm finding that I can still connect from an external SIP client. Also, I tried setting one SIP account from host=dynamic to host=ipaddr, and when that client tried to register, then Asterisk complained that the account wasn't supposed to be trying to register. My next step is also to upgrade my Asterisk itself up to the latest stable 1.6 Any other suggestions? Thanks, Adrian -- _ -- 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] Resetting Marker Bits
Anyone have any idea on how to force marker bits on in RTP ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 10 June 2009 14:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Resetting Marker Bits (resend as apparently I was blocked) Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client - A*k1 - A*k2 - PSTN Provider/Gradwell - O2 - Mobile - the SIP client dials on O2 mobile, call goes out to A*1. - A*1 Dials out to A*k2 as A*k2 is the gateway to PSTN providers and normal office phones. - A*k2 dials some local Cisco phones, then on no answer plays an audio file, so call is ANSWERED. - A*k2 then Dials out to gradwell, to a mobile phone number. - Gradwell takes the call, routes it via PSTN. My problem, is that at the point where the O2 mobile accepts the call, I get one-way audio. (SIP Client outbound, nothing inbound). Tracing the RTP stream all the way back, I can see that audio makes it all the way to the SIP Client. However, we notice that at the point where the O2 mobile answers, the TIME= value of the packet jumps significantly, say from 119248 to 1518324408. Talking to the sip client developer, they say that I need to enable SIP Markers on the server (I guess A*k2), so that if the stream source changes then the timers are reset. Does this sound right, and if so, how do I do that ? I am running an older load on A*K2 of 1.4.18, and 1.4.15svn (privately compiled to add an extra codec) on A*k1. I can look into upgrading these, but the developer thinks it's just a missing config on Asterisk. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Resetting Marker Bits
Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client - A*k1 - A*k2 - PSTN Provider/Gradwell - O2 - Mobile - the SIP client dials on O2 mobile, call goes out to A*1. - A*1 Dials out to A*k2 as A*k2 is the gateway to PSTN providers and normal office phones. - A*k2 dials some local Cisco phones, then on no answer plays an audio file, so call is ANSWERED. - A*k2 then Dials out to gradwell, to a mobile phone number. - Gradwell takes the call, routes it via PSTN. My problem, is that at the point where the O2 mobile accepts the call, I get one-way audio. (SIP Client outbound, nothing inbound). Tracing the RTP stream all the way back, I can see that audio makes it all the way to the SIP Client. However, we notice that at the point where the O2 mobile answers, the TIME= value of the packet jumps significantly, say from 119248 to 1518324408. Talking to the sip client developer, they say that I need to enable SIP Markers on the server (I guess A*k2), so that if the stream source changes then the timers are reset. Does this sound right, and if so, how do I do that ? I am running an older load on A*K2 of 1.4.18, and 1.4.15svn (privately compiled to add an extra codec) on A*k1. I can look into upgrading these, but the developer thinks it's just a missing config on Asterisk. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Resetting Marker Bits
(resend as apparently I was blocked) Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client - A*k1 - A*k2 - PSTN Provider/Gradwell - O2 - Mobile - the SIP client dials on O2 mobile, call goes out to A*1. - A*1 Dials out to A*k2 as A*k2 is the gateway to PSTN providers and normal office phones. - A*k2 dials some local Cisco phones, then on no answer plays an audio file, so call is ANSWERED. - A*k2 then Dials out to gradwell, to a mobile phone number. - Gradwell takes the call, routes it via PSTN. My problem, is that at the point where the O2 mobile accepts the call, I get one-way audio. (SIP Client outbound, nothing inbound). Tracing the RTP stream all the way back, I can see that audio makes it all the way to the SIP Client. However, we notice that at the point where the O2 mobile answers, the TIME= value of the packet jumps significantly, say from 119248 to 1518324408. Talking to the sip client developer, they say that I need to enable SIP Markers on the server (I guess A*k2), so that if the stream source changes then the timers are reset. Does this sound right, and if so, how do I do that ? I am running an older load on A*K2 of 1.4.18, and 1.4.15svn (privately compiled to add an extra codec) on A*k1. I can look into upgrading these, but the developer thinks it's just a missing config on Asterisk. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call quality - how to debug
Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, Thanks Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality - how to debug
Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality - how to debug
I don't need QoS. The voice network here is seperated from the PC LAN physically (2 separate switches), by design. So theres no web browsing etc on that 2mb circuit. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality - how to debug
Scratch that, my inventory tool says the system has 256Mb not 1Gb. I wonder if a memory upgrade would help it out... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 02 June 2009 14:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality - how to debug
Yeah, I know, but when I last tried an upgrade to 1.4.18 it broke the whole IAX connectivity and I was forced to drop back. I'll go: 1) Memory upgrade first 2) Clone the machine, and upgrade to latest 1.4.x However - my question would still stand, how exactly would I be able to debug whats going on in the RTP stream? And why its stuttering (sometimes halfway through a call). Any tips or tricks for actually debugging within Asterisk ? Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick Hartman Sent: 02 June 2009 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Do you have any idea the number of bugs that have been fixed since 1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this. On 06/02/2009 08:58 AM, Adrian Marsh wrote: Hi, It's a 2mb dedicated leased fibre line, with50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality - how to debug
Hi Dave, You're quite right, it's a dedicated down and uplink to my ISP, and Gradwell also has fibre connection into that ISP (so short hop to them) The reason I don't think it's the fiber link, is that Asterisk recorded the conversation as two channels. IN (from Gradwell), and OUT (from the Cisco phone, that's on the same LAN as the asterisk server). And I hear distortion on both sides, at the same time. As thats what asterisk hears, and that part of the call is a same-LAN RTP stream, pre-ISP, then that's why I don't think it's the IAX link. That said, I've not got complaints from users making internal calls. So my thinking was maybe its an IAX/SIP conversion thing As a test, I've switched my account, and the problem account to inbound SIP, to see if that makes a difference. That makes it 100% SIP. Next step, memory upgrade and the A*k upgrade. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 16:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug Unless I've misunderstood and you're not running ANYTHING but voice over that internet uplink? snip So theres no web browsing etc on that 2mb circuit. /snip In which case, I stand corrected and you don't need QOS. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
Re: [asterisk-users] Call quality - how to debug
Hi Steve, Mainly because, if it were a CPU utilisation issue, then putting an extra load on the server because of tcpdump isn't going to help. If I go that route then I'll port mirror on the switch. But thanks for the reply, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 16:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. Why not just tcpdump on the asterisk box then load it into wireshark? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Domains
Thanks Dave and Geraint for the reply, I'll be really specific: What does the realm= and the domain= in sip.conf actually control?? And how do they relate into Guest INVITE messages ? Dave - yes you've got it pretty right: I'm basically dialling a number (5550) from a sip client to server B and having the call passed onto server A via guest INVITE (at least I'm expecting it to be as a guest, but not so sure that's happening). If I register (to B) as sip client 2001, call 1 suceeds. 2001 is defined on server B, but NOT on server A If I regsiter (to B) as sip client 2000, call 2 fails. 2000 is defined on both servers. If I turn on sip set debug ip on server A, I don't see anything *anything* for the second call. However a tcpdump does show the incoming INVITE. The only obvious difference is that 2000 is actually defined on server A. So I think that an authentication challenge is happening. If I remove the definition on server A for client 2000, then the second call behaves just as the first. The extensions.conf line for server B is: Exten = 5550,1,Dial(5...@servera.company.com) What this is telling me so far, is that if my server gets an INVITE and the client reports its username as an ID that happens to be defined on my server, then a challenge will be sent. Now that makes perfect sense, except in my case server B is acting as an intermiediatory, and I would of thought that server A would see that (via the Domain configs) - hence my questions on Domains. For the time being, I'm ignoring why the debug on Server A shows nothing, not even the inbound invite on the second call. Thanks Adrian From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Walker Sent: 27 May 2009 22:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Domains I read through your question a couple of times. Basically you have server A which has extension 2000 and 5550. Server B has extension 2000 and 2001. You configure a (soft)phone as extension 2001 and dial 5550 which succeeds but you dial 2000 and the call fails. Have you tried turning up the debug verbosity in the console and watching the call flow on Server B? I don't know what would prompt Server B to try passing the call to Server A but that should become apparent in the debug information. If the 'domain' you are referring too his the FQDN then that has nothing to do with the price of bread as far as I can tell. Noone can give me a clue on this ? How Domains are used within Asterisk ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 26 May 2009 12:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Domains Hi, I'm trying to understand an issue I'm seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then the call works, and is placed through to server As logic successfully. But if I call in as client 2000, then the call fails, server A shows no log at all of the call (even a sip set debug ip ip showed nothing - though tcpdump did show the inbound invite). However if I remove the definition of client 2000 from server A, then the call succeeds. So I think that for a defined account server A is wanting to challenge for a password, even though the inbound call is not a local account - hence my trying now to understand if and how Asterisk uses Domains. If I define a serverA.company.com domain on server A, will it ignore the challenge for an INVITE coming from server B ?? Thanks Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Converting Cisco 7961 to SIP
I'd like to see that link too! I use Cisco 7940s at the moment, and would like to see how to hook them into AD -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: 26 May 2009 15:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Please do! Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Ahh I see. In response to your other question about the auto-provisioning of Cisco phones, I wrote some scripts that work against an active directory and setup the phones automagically. I'll send the link your way if you'd like. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Did not mean to infer they don't perform wonderfully with Asterisk. By hack I meant that Cisco does not offer any official support for them on Asterisk. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Cory, Precisely what do you mean by 'Anything other than Callmanager will essentially be a hack'? I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP image. They're not 'hacked', they're set up properly against the Cisco provided SIP image and are rock-solid stable. I would pit them against any of the cheaper model SIP phones any time, any place, any day. I've written scripts to do nearly everything that call manager can do without paying hundreds of dollars per user for the call manager software. Just about the only thing they can't do at the moment is BLF because they require SIP over TCP to handle SIP messages about BLF status, something that I'm not willing to implement just yet. In the past, Cisco phones have had a bad rap as not being usable outside of a call manager environment. That's just not the case. -Dave -Original
Re: [asterisk-users] Domains
Noone can give me a clue on this ? How Domains are used within Asterisk ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 26 May 2009 12:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Domains Hi, I'm trying to understand an issue I'm seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then the call works, and is placed through to server As logic successfully. But if I call in as client 2000, then the call fails, server A shows no log at all of the call (even a sip set debug ip ip showed nothing - though tcpdump did show the inbound invite). However if I remove the definition of client 2000 from server A, then the call succeeds. So I think that for a defined account server A is wanting to challenge for a password, even though the inbound call is not a local account - hence my trying now to understand if and how Asterisk uses Domains. If I define a serverA.company.com domain on server A, will it ignore the challenge for an INVITE coming from server B ?? Thanks Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Domains
Hi, I'm trying to understand an issue I'm seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then the call works, and is placed through to server As logic successfully. But if I call in as client 2000, then the call fails, server A shows no log at all of the call (even a sip set debug ip ip showed nothing - though tcpdump did show the inbound invite). However if I remove the definition of client 2000 from server A, then the call succeeds. So I think that for a defined account server A is wanting to challenge for a password, even though the inbound call is not a local account - hence my trying now to understand if and how Asterisk uses Domains. If I define a serverA.company.com domain on server A, will it ignore the challenge for an INVITE coming from server B ?? Thanks Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] inbound SIP funnies
Hi, I've a few working asterisk servers, all seeing the same symptom, but they are all based on the same configs. A SIP inbound INVITE message is coming in to an extension (not a peer) eg 5...@ourserver.com A tcpdump clearly shows the INVITE coming in, but asterisk seems to be ignoring it (theres no reply outbound packet). All the source/dest IPs and ports look good. A sip set debug trace ip sourceip is blank, showing nothing at all. The sip.conf default context is incoming_pstn. The incoming_pstn context is: [incomming_pstn] include = local-UK include = local-US include = test_numbers and [test_numbers] includes: exten = 555,1,Answer(0) ; Pick up phone instantly exten = 555,n,Playback(vq51) ; Let them know what's going on exten = 555,n,Playback(vq20) exten = 555,n,Goto(default,555,3) ; repeat So as far as I can tell, we should be accepting the connection and playing the voicefile (yup - I know this would be open to the internet, that's the intention). Sip.conf also has: allowexternalinvites=yes allowexternaldomains=yes so it should be working I think... This is a 1.4.15 based asterisk Thanks Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Proxying from one server to another
Hi All, I'm trying to find a software package to do the following sip proxy work: I've an A*k server A that needs to be decommissioned, from the USA, and replaced by server B, in the UK. Both servers are on public internet IPs. Whilst the client migration happens, I want to divert all the Register traffic from Server A to Server B to catch any clients still left out there. Unfortunately, the original Clients were configured with static IPs instead of DNS names for the SIP Registrar, so I have to proxy Server A until all the clients have been updated (which might be a long time). Obviously A*k itself wont do this (as far as I know). I've looked at siproxyd and party-sip, but with no success so far. I've also tried using IPtables to redirect at the IP level, but the public IP ranges seem to stop me from achieving this. It works in my local-lan testing, but not on the public servers. Any ideas? Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proxying from one server to another
Hi David, Thanks for the reply. That's pretty much what I've already tried, but with no luck on the production machines. In testing it worked, but the public IPs and single NICs were causing issues (we believe) So I was looking for a proxy-type solution. Adrian From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 13 May 2009 15:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Proxying from one server to another Redirect traffic with iptables like this: Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to NEW_PUBLIC_IP I'm not sure if this will work for SIP. You may need the proxy to change info in the sip messages between server and client. --Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Wednesday, May 13, 2009 8:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Proxying from one server to another Hi All, I'm trying to find a software package to do the following sip proxy work: I've an A*k server A that needs to be decommissioned, from the USA, and replaced by server B, in the UK. Both servers are on public internet IPs. Whilst the client migration happens, I want to divert all the Register traffic from Server A to Server B to catch any clients still left out there. Unfortunately, the original Clients were configured with static IPs instead of DNS names for the SIP Registrar, so I have to proxy Server A until all the clients have been updated (which might be a long time). Obviously A*k itself wont do this (as far as I know). I've looked at siproxyd and party-sip, but with no success so far. I've also tried using IPtables to redirect at the IP level, but the public IP ranges seem to stop me from achieving this. It works in my local-lan testing, but not on the public servers. Any ideas? Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Codecs
All, I think we've found what was blocking us. It seems that SElinux, for some unknown reason, didn't like the AMR codec, and did something to block it. Set that to passive, and the problem goes away... Would still like to learn more about asterisk codec translation though, if anyone has any pointers. Adrian From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 07 May 2009 09:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding Codecs Hi All, My theory on the codec translation deepens: Doing a core show translation on the A1 server (working) I get: g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr g723- ---- -- -- -- -- - gsm- -222 21 3- - 11 2- 45 ulaw- 2-12 21 3- - 11 2- 45 alaw- 21-2 21 3- - 11 2- 45 g726aal2- 222- 21 3- - 11 1- 45 adpcm- 2222 -1 3- - 11 2- 45 slin- 1111 1- 2- - 10 1- 44 lpc10- 2222 21 -- - 11 2- 45 g729- ---- -- -- -- -- - speex- ---- -- -- -- -- - ilbc- 2222 21 3- -- 2- 45 g726- 2221 21 3- - 11 -- 45 g722- ---- -- -- -- -- - amr- 13 13 13 1313 1214- - 22 13- - But on the new server it gives: g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr g723- ---- -- -- -- -- - gsm- -222 21 2- - 11 2- - ulaw- 2-12 21 2- - 11 2- - alaw- 21-2 21 2- - 11 2- - g726aal2- 222- 21 2- - 11 1- - adpcm- 2222 -1 2- - 11 2- - slin- 1111 1- 1- - 10 1- - lpc10- 2222 21 -- - 11 2- - g729- ---- -- -- -- -- - speex- ---- -- -- -- -- - ilbc- 2222 21 2- -- 2- - g726- 2221 21 2- - 11 -- - g722- ---- -- -- -- -- - amr- ---- -- -- -- -- - So where are the codec translations set? Thanks Adrian From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 06 May 2009 18:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding Codecs Forgot to add: sip.conf for both A1 and A2 has the following global codec definitions: disallow=all allow=clear allow=amr allow=ulaw allow=alaw The Asterisk build is a private build that adds the clear and AMR codec setups. The two servers are running Fedora, though A1s on 6 and A2s on 10. I cant see why that would make a difference though. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 06 May 2009 17:53 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Understanding Codecs Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and b A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 14958 Adding codec 0x2000 (amr) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call
Re: [asterisk-users] Understanding Codecs
Ah... ok thanks for that. In the end it was an SElinux problem. But I was curious as to if I was missing some config somewhere. This clears that up. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: 07 May 2009 15:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding Codecs On Thursday 07 May 2009 03:33:14 Adrian Marsh wrote: So where are the codec translations set? I assume you're talking about the numbers within the table? They're calculated at runtime, based upon shortest possible path (in terms of time) from one codec to another. Most codecs translate only to signed linear audio, so the translation table tends to be rather simple. Ulaw to alaw is a simple table lookup, which is why it tends to be very fast. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Proxying comparison
Hi All, Looking to gauge some opinions on redirect/proxy software. I've two existing A*k servers out on the 'net. I need to redirect the traffic going to those two servers, over to a new 3rd one. Unfortunately, when the servers and clients were built, they used hardcoded IPs, rather than DNS, so a simple DNS update wont work. So I'm looking at IP redirect, or SIP proxying as options. In my lab tests here, using iptables and forwarding the IP packets seemed to work really well. However on the hosted servers, it seems I'm unable to do this with iptables, due to one reason or another on the hosted platforms (Plesk being the main issue). So - for proxying then, which is the most simplest proxy server to setup? I've been playing with siproxd today, and wondering about OpenSER. But I've not used SIP proxys before. I just need a very simple redirect. Anything inbound SIP redirect over to the new server. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Codecs
Hi All, My theory on the codec translation deepens: Doing a core show translation on the A1 server (working) I get: g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr g723- ---- -- -- -- -- - gsm- -222 21 3- - 11 2- 45 ulaw- 2-12 21 3- - 11 2- 45 alaw- 21-2 21 3- - 11 2- 45 g726aal2- 222- 21 3- - 11 1- 45 adpcm- 2222 -1 3- - 11 2- 45 slin- 1111 1- 2- - 10 1- 44 lpc10- 2222 21 -- - 11 2- 45 g729- ---- -- -- -- -- - speex- ---- -- -- -- -- - ilbc- 2222 21 3- -- 2- 45 g726- 2221 21 3- - 11 -- 45 g722- ---- -- -- -- -- - amr- 13 13 13 1313 1214- - 22 13- - But on the new server it gives: g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr g723- ---- -- -- -- -- - gsm- -222 21 2- - 11 2- - ulaw- 2-12 21 2- - 11 2- - alaw- 21-2 21 2- - 11 2- - g726aal2- 222- 21 2- - 11 1- - adpcm- 2222 -1 2- - 11 2- - slin- 1111 1- 1- - 10 1- - lpc10- 2222 21 -- - 11 2- - g729- ---- -- -- -- -- - speex- ---- -- -- -- -- - ilbc- 2222 21 2- -- 2- - g726- 2221 21 2- - 11 -- - g722- ---- -- -- -- -- - amr- ---- -- -- -- -- - So where are the codec translations set? Thanks Adrian From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 06 May 2009 18:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding Codecs Forgot to add: sip.conf for both A1 and A2 has the following global codec definitions: disallow=all allow=clear allow=amr allow=ulaw allow=alaw The Asterisk build is a private build that adds the clear and AMR codec setups. The two servers are running Fedora, though A1s on 6 and A2s on 10. I cant see why that would make a difference though. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 06 May 2009 17:53 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Understanding Codecs Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and b A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 14958 Adding codec 0x2000 (amr) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to B, it only offers amr: [May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 15554 Adding codec 0x2000 (amr) to SDP Adding non-codec 0x1 (telephone-event) to SDP Its not building ulaw or alaw into its list. Server B doesn't support AMR, so rejects the call. (I've no idea about the 0x4000 error - but I see it on both the good and bad servers, so I don't think its related). The odd thing is that the sip.conf files for A1 and A2 are exactly the same (save IP info). The build of the Asterisk server is from a 1.4.15 private build
[asterisk-users] Understanding Codecs
Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and b A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 14958 Adding codec 0x2000 (amr) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to B, it only offers amr: [May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 15554 Adding codec 0x2000 (amr) to SDP Adding non-codec 0x1 (telephone-event) to SDP Its not building ulaw or alaw into its list. Server B doesn't support AMR, so rejects the call. (I've no idea about the 0x4000 error - but I see it on both the good and bad servers, so I don't think its related). The odd thing is that the sip.conf files for A1 and A2 are exactly the same (save IP info). The build of the Asterisk server is from a 1.4.15 private build to add AMR, but, it's the same source built on both A1 and A2. I'm trying to figure out why A2 isnt offering ulaw and alaw. The codec seems ok, and is listed in the show codecs: 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 8192 (1 13) (0x2000) audioamr (AMR) But I cant see why its not transcoding across to ulaw/alaw. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Codecs
Forgot to add: sip.conf for both A1 and A2 has the following global codec definitions: disallow=all allow=clear allow=amr allow=ulaw allow=alaw The Asterisk build is a private build that adds the clear and AMR codec setups. The two servers are running Fedora, though A1s on 6 and A2s on 10. I cant see why that would make a difference though. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 06 May 2009 17:53 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Understanding Codecs Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and b A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 14958 Adding codec 0x2000 (amr) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to B, it only offers amr: [May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 15554 Adding codec 0x2000 (amr) to SDP Adding non-codec 0x1 (telephone-event) to SDP Its not building ulaw or alaw into its list. Server B doesn't support AMR, so rejects the call. (I've no idea about the 0x4000 error - but I see it on both the good and bad servers, so I don't think its related). The odd thing is that the sip.conf files for A1 and A2 are exactly the same (save IP info). The build of the Asterisk server is from a 1.4.15 private build to add AMR, but, it's the same source built on both A1 and A2. I'm trying to figure out why A2 isnt offering ulaw and alaw. The codec seems ok, and is listed in the show codecs: 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 8192 (1 13) (0x2000) audioamr (AMR) But I cant see why its not transcoding across to ulaw/alaw. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ENUM lookup
I'll be sure to post back if I think of anything as I go Adrian Marsh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: 14 August 2008 14:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ENUM lookup On Thursday 14 August 2008 07:33:11 Brian J. Murrell wrote: On Thu, 2008-08-14 at 14:15 +0200, Klaus Darilion wrote: Use the ENUMLOOKUP function, e.g.: And take note that it's very naive. See my previous posting for an enum AGI that is more intelligent. The only thing it does not do that I would like to add is give up on the DNS lookup much earlier than it does if a DNS server is unresponsive. If you'd like to give a suggestion on how to make the ENUMLOOKUP function more useful, I'm all ears. Sometimes the issue is that the people who are most qualified to make the dialplan functions more useful aren't in a position to do anything about it (either because they aren't C programmers or because they aren't ENUM users). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ENUM lookup
Thanks Brian, I do remember seeing references to that AGI, but I've not used AGI much yet either so was looking for something simple to setup (hence the original SIPbroker config). Will try to find it though. Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Murrell Sent: 14 August 2008 13:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ENUM lookup On Thu, 2008-08-14 at 14:15 +0200, Klaus Darilion wrote: Use the ENUMLOOKUP function, e.g.: And take note that it's very naive. See my previous posting for an enum AGI that is more intelligent. The only thing it does not do that I would like to add is give up on the DNS lookup much earlier than it does if a DNS server is unresponsive. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ENUM lookup
Hi All, For a 1.4 version asterisk, whats the recommended mechanism for dialling with ENUM lookup? At the moment I user SIPbroker, but am getting tired of it hanging on certain numbers, so I was thinking about implementing it myself. I've seen various vo-ip.info pages (http://www.voip-info.org/wiki/view/Asterisk+cmd+EnumLookup) talking about the func ENUMLOOKUP instead of EnumLookup Application, but then I'll need to implement my own logic around this right?? Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem controlling dialplan order
Hi All,
On a 1.4.15 system, I've a context as below, where I need to catch some
specific US ranges and dial direct via SIP rather than a PSTN trunk.
But the logic always goes via the International Trunk and I cant see
why...
[local]
exten = _00165011091[45]0-9],1,NoOp(I AM HERE)
exten = _00165011091[45]0-9],n,Macro(setcli)
exten = _00165011091[45]0-9],n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00165011091[45]0-9],n,Hangup
. (same context)
Catch local (UK) numbers
exten = _0[1-9]X.,1,NoOp(Dialling UK number)
exten = _0[1-9]X.,n,Macro(setcli)
exten = _0[1-9]X.,n(jumpdial),Dial(SIP/+44${EXTEN:[EMAIL PROTECTED])
exten = _0[1-9]X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)
exten = _0[1-9]X.,n+101,Busy
;Catch any (00xx) numbers
exten = _00X.,1,NoOp(Dialling International number)
exten = _00X.,n,Macro(setcli)
exten = _00X.,n(jumpdial),Dial(SIP/+${EXTEN:[EMAIL PROTECTED])
exten = _00X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)
exten = _00X.,n+101,Busy
I've tried putting the Catch codes above into a sub-context, and then
put an include into the [local], but it still dials via the Catch
international...
The odd thing is that in either, the show dialplan seems to suggest the
correct order :
'_00165011091[45]0-9]' = 1. NoOp(I AM HERE)
[pbx_config]
2. Macro(setcli)
[pbx_config]
3. Dial(SIP/${EXTEN:[EMAIL PROTECTED]) [pbx_config]
4. Hangup()
[pbx_config]
(some others)
'_00X.' =1. NoOp(Dialling International number)
[pbx_config]
2. Macro(setcli)
[pbx_config]
[jumpdial] 3. Dial(SIP/+${EXTEN:[EMAIL PROTECTED])
[pbx_config]
104. Dial(${TRUNK}/${EXTEN}||Wr)
[pbx_config]
206. Busy()
[pbx_config]
The page at voip-info isn't too clear in the differences between 1.2 and
1.4
(http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sort
ing) so I'm not sure where I've gone wrong.
Adrian Marsh
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Re: [asterisk-users] problem controlling dialplan order
Oh for
Stared at that for ages not seeing it
Thanks Felippe...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Felippe
Silvestre
Sent: 07 August 2008 17:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] problem controlling dialplan order
Try this:
[local]
exten = _00165011091[45][0-9],1,NoOp(I AM HERE)
exten = _00165011091[45][0-9],n,Macro(setcli)
exten = _00165011091[45][0-9],n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00165011091[45][0-9],n,Hangup
The [ before 0-9] is needed.
Felippe Silvestre
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: Thursday, August 07, 2008 07:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] problem controlling dialplan order
Hi All,
On a 1.4.15 system, I've a context as below, where I need to
catch some specific US ranges and dial direct via SIP rather than a PSTN
trunk. But the logic always goes via the International Trunk and I cant
see why...
[local]
exten = _00165011091[45]0-9],1,NoOp(I AM HERE)
exten = _00165011091[45]0-9],n,Macro(setcli)
exten =
_00165011091[45]0-9],n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00165011091[45]0-9],n,Hangup
. (same context)
Catch local (UK) numbers
exten = _0[1-9]X.,1,NoOp(Dialling UK number)
exten = _0[1-9]X.,n,Macro(setcli)
exten = _0[1-9]X.,n(jumpdial),Dial(SIP/+44${EXTEN:[EMAIL PROTECTED])
exten = _0[1-9]X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)
exten = _0[1-9]X.,n+101,Busy
;Catch any (00xx) numbers
exten = _00X.,1,NoOp(Dialling International number)
exten = _00X.,n,Macro(setcli)
exten = _00X.,n(jumpdial),Dial(SIP/+${EXTEN:[EMAIL PROTECTED])
exten = _00X.,jumpdial+101,Dial(${TRUNK}/${EXTEN},,Wr)
exten = _00X.,n+101,Busy
I've tried putting the Catch codes above into a sub-context, and
then put an include into the [local], but it still dials via the Catch
international...
The odd thing is that in either, the show dialplan seems to
suggest the correct order :
'_00165011091[45]0-9]' = 1. NoOp(I AM HERE)
[pbx_config]
2. Macro(setcli)
[pbx_config]
3. Dial(SIP/${EXTEN:[EMAIL PROTECTED])
[pbx_config]
4. Hangup()
[pbx_config]
(some others)
'_00X.' =1. NoOp(Dialling International number)
[pbx_config]
2. Macro(setcli)
[pbx_config]
[jumpdial] 3. Dial(SIP/+${EXTEN:[EMAIL PROTECTED])
[pbx_config]
104. Dial(${TRUNK}/${EXTEN}||Wr)
[pbx_config]
206. Busy()
[pbx_config]
The page at voip-info isn't too clear in the differences between
1.2 and 1.4
(http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sort
ing) so I'm not sure where I've gone wrong.
Adrian Marsh
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Re: [asterisk-users] IAX to work on two ports: 4569 and 4570
Why would you need to to that anyway? Just set them to one port, but use different contexts to handle the inbound traffic differently. Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: 25 July 2008 14:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX to work on two ports: 4569 and 4570 On Friday 25 July 2008 05:53:38 bilal ghayyad wrote: How to let my Asterisk work able to deal with two kind of IAX channels, one work on 4569 and one work on 4570 and able to receive and send calls on these two UDP ports, depends on the destination. There really isn't any good way. The IAX2 channel will only bind to a single port. You could start a secondary Asterisk server on the same machine and pass traffic through, or you could use firewalling rules to divert from port 4570 to 4569. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reinvites and SIP/RTP
Hi All, When I use re-invite, does the Asterisk server stay in the SIP conversation, and just RTP traffic diverts, or does the SIP transfer away from the A*k server too ? Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't make asterisk work...how to test?
Most SIP clients have a logging ability.. you can use those.. but turning on debug on the server is the best mechanism, as its whats going on there that counts. sip set debug options And if you want to get really into the lower levels, then tcpdump will let you capture the packets for offline analysis in wireshark. Nmaping against locahost wont tell you much other than an app has 5060 open.. it wont tell you if firewalls are blocking things, or if NAT is an issue. Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of D. Dante Lorenso Sent: 20 June 2008 03:14 To: 'Asterisk Users List' Subject: [asterisk-users] Can't make asterisk work...how to test? All, I've put a new asterisk server at another location and can't seem to get it working. What's the best strategy to debug connections? I'm doing inbound SIP only and have installed the server in the same way as I did on my DEV server. Running an nmap on localhost shows the port listening: -- [asterisk]/ nmap -sU localhost Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2008-06-19 21:12 CDT Interesting ports on localhost.localdomain (127.0.0.1): Not shown: 1476 closed ports PORT STATE SERVICE ... 5060/udp open|filtered sip ... -- [planet]/etc/asterisk nmap -sU localhost Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2008-06-19 20:11 CDT Interesting ports on localhost.localdomain (127.0.0.1): Not shown: 1484 closed ports PORT STATE SERVICE ... 5060/udp open|filtered sip ... -- Is there a command-line tool I can run that will attempt a SIP connection to a SIP server and provide some diagnostics about whether it could authenticate or even connect? -- Dante ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference withabout 12 other people from around the USA. Bandwidth issues aside, willthis work or will all the different latencies cause issues? Yea I know,I could just try it and find out but it is going to take alot of timeto get everyones schedule to line up, I don't want to go through thetrouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Default ringtone
Hi All, I've trying to force on the ringtone generated for outbound calls with Dial,r but want the tone to be the UK standard. I use Zaptel, but don't have any E1/T1 cards at all (am completely IP based). So I don't think zaptel.conf will come into this (am I right??) I've tried editing zapel.conf anyway, and changed loadzone and defaultzone to =uk I've read through zapara.conf, but cant see a ringtone definition in there. Despite these changes and a restart of zaptel and asterisk via /etc/init.d, I still hear a US ringing sound. So what did I miss? Also, is it possible to generate different ringtones based on dialplan? Eg, if I dial out to a UK number, use the UK ring, but for US use a US one ? In the past I've tried using playtone(), but that stops immediately that the our IP-carrier picks up the call. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default ringtone
Hmmm.. Well indications.conf does have: country=uk But I've definitly just hearing a long-tone tone, long break, long tone But the file is set to: [uk] description = United Kingdom ringcadence = 400,200,400,2000 ; These are the official tones taken from BT SIN350. The actual tones ; used by BT include some volume differences so sound slightly different ; from Asterisk-generated ones. dial = 350+440 Any idea why? Thanks Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 05 June 2008 15:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default ringtone Adrian Marsh wrote: Hi All, I've trying to force on the ringtone generated for outbound calls with Dial,r but want the tone to be the UK standard. I use Zaptel, but don't have any E1/T1 cards at all (am completely IP based). So I don't think zaptel.conf will come into this (am I right??) I've tried editing zapel.conf anyway, and changed loadzone and defaultzone to =uk I've read through zapara.conf, but cant see a ringtone definition in there. Despite these changes and a restart of zaptel and asterisk via /etc/init.d, I still hear a US ringing sound. So what did I miss? Also, is it possible to generate different ringtones based on dialplan? Eg, if I dial out to a UK number, use the UK ring, but for US use a US one ? In the past I've tried using playtone(), but that stops immediately that the our IP-carrier picks up the call. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users indications.conf is the file you want to edit :) It defines what ringtones and other indication signals to use. -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default ringtone
So I wonder, is it asterisk itself generating the tones in Dial(), or does it comefom the psedo zaptel driver that generates it ?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 05 June 2008 16:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default ringtone Adrian Marsh wrote: Hmmm.. Well indications.conf does have: country=uk But I've definitly just hearing a long-tone tone, long break, long tone But the file is set to: [uk] description = United Kingdom ringcadence = 400,200,400,2000 ; These are the official tones taken from BT SIN350. The actual tones ; used by BT include some volume differences so sound slightly different ; from Asterisk-generated ones. dial = 350+440 Any idea why? Thanks Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 05 June 2008 15:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default ringtone Adrian Marsh wrote: Hi All, I've trying to force on the ringtone generated for outbound calls with Dial,r but want the tone to be the UK standard. I use Zaptel, but don't have any E1/T1 cards at all (am completely IP based). So I don't think zaptel.conf will come into this (am I right??) I've tried editing zapel.conf anyway, and changed loadzone and defaultzone to =uk I've read through zapara.conf, but cant see a ringtone definition in there. Despite these changes and a restart of zaptel and asterisk via /etc/init.d, I still hear a US ringing sound. So what did I miss? Also, is it possible to generate different ringtones based on dialplan? Eg, if I dial out to a UK number, use the UK ring, but for US use a US one ? In the past I've tried using playtone(), but that stops immediately that the our IP-carrier picks up the call. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users indications.conf is the file you want to edit :) It defines what ringtones and other indication signals to use. Sorry I don't, wish I could be of more help. I'll see what I can dig up -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logical AND
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten = s,n,NoOp(${PSTN_NUM})
exten = s,n,ExecIf( $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten = s,n,NoOp(${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d0f518,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d0f518, 0
1|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d0f518,
0010123456789) in new stack
It should evaluate PSTN_NUM and add a 001 only if: PSTN_NUM is 10 digits
long and doesn't start with a 0.
However, from the debug it's being changed, even though the first test
operator logically is 0. If seems as though the isnt being applied.
Any ideas?
Thanks
Adrian
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[asterisk-users] Logical AND (resent due to bounces)
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten = s,n,NoOp(${PSTN_NUM})
exten = s,n,ExecIf( $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten = s,n,NoOp(${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d0f518,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d0f518, 0
1|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d0f518,
0010123456789) in new stack
It should evaluate PSTN_NUM and add a 001 only if: PSTN_NUM is 10 digits
long and doesn't start with a 0.
However, from the debug it's being changed, even though the first test
operator logically is 0. If seems as though the isnt being applied.
Any ideas?
Thanks
Adrian
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Re: [asterisk-users] Transfer
Thanks Sherwood, But how do I send back a 302, once I'm already in the dialplan (hasn't asterisk already sent back a 200 OK by this point??) Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 23 May 2008 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Adrian Marsh wrote: Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send the call back to the network, which in turn then routed the call appropriately. It added a transfer-number into the SS7 headers so that the originating number, dialed number and transfer number all stayed to specs, and everyone was happy. In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to have at least the control packets go via my SIP server), and use a Dial out to the far end. So - is there a way of handing the call back to the network in asterisk ? My detailed problem is this: When a call comes in, I want to send it onto users mobiles, if I hairpin the call that's OK, except the CLI needs to be that of the originator (from the USERS point of view) so they can decide if they want to accept the call. Here in the UK, this is where the issues begin... the carriers here don't like it if your sending CLI for other countries, that don't match what they think they should receive from that connecting carrier. Eg, if a call coming to them is 13 digits, but they only expect 11 from that carrier, then they cut the digits. This turns a US originated call into a Southampton UK originated call! So I was hoping that handing the call back to the network in the traditional sense would make it their problem and not mine... lol Thanks, Adrian -- -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.google.com/search?hl=enq=asterisk+302+redirect+sipbtnG=Sear ch I believe you're looking for a 302 Redirect? Sorry if you're not, but that sounds like what you want ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer
Thanks Sherwood, But how do I send back a 302, once I'm already in the dialplan (hasn't asterisk already sent back a 200 OK by this point??) Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 23 May 2008 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Adrian Marsh wrote: Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send the call back to the network, which in turn then routed the call appropriately. It added a transfer-number into the SS7 headers so that the originating number, dialed number and transfer number all stayed to specs, and everyone was happy. In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to have at least the control packets go via my SIP server), and use a Dial out to the far end. So - is there a way of handing the call back to the network in asterisk ? My detailed problem is this: When a call comes in, I want to send it onto users mobiles, if I hairpin the call that's OK, except the CLI needs to be that of the originator (from the USERS point of view) so they can decide if they want to accept the call. Here in the UK, this is where the issues begin... the carriers here don't like it if your sending CLI for other countries, that don't match what they think they should receive from that connecting carrier. Eg, if a call coming to them is 13 digits, but they only expect 11 from that carrier, then they cut the digits. This turns a US originated call into a Southampton UK originated call! So I was hoping that handing the call back to the network in the traditional sense would make it their problem and not mine... lol Thanks, Adrian -- -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.google.com/search?hl=enq=asterisk+302+redirect+sipbtnG=Sear ch I believe you're looking for a 302 Redirect? Sorry if you're not, but that sounds like what you want ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logical AND
Hi Steve,
I can see what yours does, but I still get the same end result (even
though theres only a single 0 result now)
:
exten = s,n,ExecIf( $[ $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ] ] |Set|PSTN_NUM=001${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d9a9a0,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d9a9a0, 0
|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d9a9a0,
0010123456789) in new stack
-
hello,
you should try this:
exten = s,n,ExecIf($[ $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ]]|Set|PSTN_NUM=001${PSTN_NUM})
cause the AND Operator is another thing to work, so the result at your
way look like this ExecIf(11 | ...) and with my way it looks like this
ExecIf($[11]|...) which is the right syntax for it.
best regards
steve smith
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Re: [asterisk-users] Logical AND
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: 25 May 2008 15:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Logical AND
On Sunday 25 May 2008 07:10:22 Adrian Marsh wrote:
exten = s,n,ExecIf( $[ $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ] ] |Set|PSTN_NUM=001${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d9a9a0,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d9a9a0, 0
|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d9a9a0,
0010123456789) in new stack
There's an extra space between the opening ( and the opening $[, so
the result is space-zero, which is not the same thing as zero. Any
string
that is not exactly 0 (or the empty string), such as foo, , or
0 is
true.
--
Tilghman
-
I did wonder where the extra spaces were coming from, but I thought that
was where the quotes were supposed to come into play... Well that got
it working so thanks guys..
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[asterisk-users] Transfer
Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send the call back to the network, which in turn then routed the call appropriately. It added a transfer-number into the SS7 headers so that the originating number, dialed number and transfer number all stayed to specs, and everyone was happy. In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to have at least the control packets go via my SIP server), and use a Dial out to the far end. So - is there a way of handing the call back to the network in asterisk ? My detailed problem is this: When a call comes in, I want to send it onto users mobiles, if I hairpin the call that's OK, except the CLI needs to be that of the originator (from the USERS point of view) so they can decide if they want to accept the call. Here in the UK, this is where the issues begin... the carriers here don't like it if your sending CLI for other countries, that don't match what they think they should receive from that connecting carrier. Eg, if a call coming to them is 13 digits, but they only expect 11 from that carrier, then they cut the digits. This turns a US originated call into a Southampton UK originated call! So I was hoping that handing the call back to the network in the traditional sense would make it their problem and not mine... lol Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Googles 411 services
Hi Brian, Thanks for the reply. I tried searching for your posts, but no luck. Do you have sample code I could see? Or do you just take a +101 result and move to the next in the list?? Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Murrell Sent: 17 May 2008 21:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Googles 411 services On Sat, 2008-05-17 at 18:38 +0100, Adrian Marsh wrote: All, Does anyone know of a SIP URI direct to googles 800-GOOG-411 service? Yeah, I suppose a direct SIP connection would be nice. An enum lookup shows 3 URIs listed, none of them seem to be google directly, No, they are SIP-PSTN termination services. I use them via an ENUM lookup for all of my toll-free calling since my ITSP doesn't terminate toll-free for me at no charge. and I think 1 of them fails 100%, and the remaining one fails at other random times. Yeah, they do have a random failure rate, which is why my enum macro returns all three and rolls over to alternate values if any fail. Check the archives (within the last few weeks) for more details. b, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sipbroker CLI
Hi, Can anyone confirm if calls placed via sipbroker have their NUM CLI changed by sipbroker?? I'm testing between two asterisk servers in seperate locations. When I place a call directly, the CLI is fine. When the call is placed via sipbroker lookup, the NAME stays the same, but the NUM is recieved as sipbroker. I'm trying to figure out if its being set by the sending Asterisk server SIP account, or if siproker themselves would mess with the CLI. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Googles 411 services
All, Does anyone know of a SIP URI direct to googles 800-GOOG-411 service? When I put calls via sipbroker, half the time the calls fail. An enum lookup shows 3 URIs listed, none of them seem to be google directly, and I think 1 of them fails 100%, and the remaining one fails at other random times. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SLN File Format
Hi All, Whats the SLN file format (for import/export to Audacity)? Need to avoid Sox if I can Adrian Marsh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLN File Format
My exact requirement.. to edit out some recorded hiss and then put the file back... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: 08 May 2008 15:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SLN File Format Philipp Kempgen wrote: Just out of curiosity: I can't remember when I last had to concatenate 2 sound files. So why does this always come up? IMHO it's one of those things you hardly ever need.(?) I can't remember the last time I have done that. :) Anytime I need to do something like that, I just use a nice tool like audacity ... -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco to Asterisk migration
Basic process: 1) Build the A*k server so that it has tftp installed (or another box that does) 2) Build up the SIPdefault.conf and get the firmware files in place (see Cisco docs on this, plus theres loads on the wikis). 3) Test with a single phone, change its tftp server to the asterisk. Check that : a) The firmware switches to SIP b) the phone registers to A*k and all is well. Calls can be made etc... 4) Once your happy with the A*k config and I mean ***really*** happy, then add in all the configs for the other phones (I used scripts to build mine). 5) Try a few more phones manually. But eventually just update DHCP so that the tftp server option points to the A*k server. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Femi Sent: 25 April 2008 10:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Cisco to Asterisk migration Hi Guys, I have client with a Cisco 2690 call manager solution that wants to upgrade but cannot stomach the costs of continuing with Cisco The installation will go up to 100 users The client currently has about 40 Cisco phones and would like to continue with these phones with the odd Polycom I'm looking at plugging in an Asterisk box and using the existing Cisco box as a PSTN gateway only Has anyone on the list done this? Any pitfalls or tips you would like to share? Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Background ring
3rd attempt.. get the right list... Hi All, When I hairpin calls out to some networks (eg international or mobiles), there can be a long delay until the PSTN starts sending audio ring tones back. Is there a way I can have asterisk play ringtones until the PSTN really answers?? I've looked at Playtone(), Background() Playback.. Playtone looks like it should do the job, but I still get a long silence. I think the PSTN may be sending an Accept signal back halting the Playtone(). Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Debugging DTMF
Hi All, I'm trying to debug DTMF issues I have with certain endpoint conferencing systems (external, 3rd party). On our A*k server I log DTMF, and I see that coming through in the log. What I'd like to see is what is sent onto our VoIP carrier over SIP. I can do a tcpdump of the packets, but what am I then looking for? Would it be in the RTP audio stream or within the SIP protocol?? I'm using Wireshark to decode... Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Removing Parsing /etc/asterisk/manager.conf from CLI
Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the == Parsing '/etc/asterisk/manager.conf': Found. (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) There is a very old feature request about this at http://bugs.digium.com/view.php?id=3085 but I cannot see the resolution. Mantis shows APPLICATION ERROR #801 at the end of the page... Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing Parsing /etc/asterisk/manager.conf from CLI
Anything more than 'core set verbose 1' produces this message, however verbose 1 does not display much of anything. On Thu, Apr 10, 2008 at 1:53 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Apr 09, 2008 at 11:55:13PM -0700, Adrian A wrote: Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the == Parsing '/etc/asterisk/manager.conf': Found. (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) What verbosity level do you use? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing Parsing /etc/asterisk/manager.conf from CLI
There is an OpenSER proxy in front of Asterisk which handles the clients. The script is called by OpenSER whenever a client sends a SUBSCRIBE request for MWI. It uses php to connect to Asterisk like so: fsockopen($mhost,5038, $errno, $errstr, 5) and gets the user's voicemail counts. I'm not sure how I would maintain this as a persistent connection that would live if I restart Asterisk. I'd have to detect that somehow. Adrian On Thu, Apr 10, 2008 at 12:14 AM, Stefan Reuter [EMAIL PROTECTED] wrote: Adrian A wrote: Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager (...) Maybe a better solution is to rethink your architecture. The Manager API is well suited for long running connections, so there is no need to reconnect every few seconds. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] WWW: http://www.reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP not being sent when on hold
The RTP codec 126 is a bogus RTP packet sent by Bria to maintain the NAT binding. I've identified the issue as this: Bria has an inactivity timer that is based on RTCP. Basically, if during the call there is RTCP, Bria uses it to make sure the call is still alive. Asterisk does send RTCP when call is active, but it stops when call is put on hold by Bria. The default timeout for Bria is 30 seconds, thus it disconnects the call because it has not received any RTP or RTCP during this time. I am not sure at this point which is correct implementation. Should the client not rely on RTP/RTCP when it's on hold or should Asterisk send some sort of keep alive RTP/RTCP when it knows one of the clients is on hold? On Wed, Apr 9, 2008 at 7:15 AM, Steve Langstaff [EMAIL PROTECTED] wrote: It would be interesting to see a wireshark trace of the SIP and RTP traffic during call setup and hold, to see: a) what codec 126 has been negotiated as and b) who is sourcing the unknown RTP datagram. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Adrian A *Sent:* 09 April 2008 00:55 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] RTCP not being sent when on hold Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0' In sip.conf I have rtpkeepalive=15 but that does not seem to help. Does anyone know what I can do to fix this, other than increase the timeout on Bria? Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTCP not being sent when on hold
Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0' In sip.conf I have rtpkeepalive=15 but that does not seem to help. Does anyone know what I can do to fix this, other than increase the timeout on Bria? Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UK GMT/BST settings
Hi, Anyone know what the settings in SIPDefault.cnf should be for Cisco 7940 phones this year? Came in today to find they'd all moved one hour ahead (NTP server is correct and ok). Found the day was set to 26, but on trying to change the settings to the below, my test phone isn't changing back: dst_start_month: March ; Month in which DST starts dst_start_day: 29 ; Day of month in which DST starts dst_start_day_of_week: Sun ; Day of week in which DST starts dst_start_week_of_month: 1 ; Week of month in which DST starts dst_start_time: 01 ; Time of day in which DST starts dst_stop_month: Oct ; Month in which DST stops dst_stop_day: 26 ; Day of month in which DST stops dst_stop_day_of_week: Sunday ; Day of week in which DST stops dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month dst_stop_time: 1 ; Time of day in which DST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
