Re: [asterisk-users] MYSQL queries from dial plan
On Mon, 4 Jan 2010, Neeraj Chand wrote: I currently run small scale mysql queries from the dialplan [snip] This currently takes about 4 seconds to complete. If I run two simultaneous queries, this goes up to about 9 seconds for both queries to complete. 4 seconds for a query is a bit extreme. 9 seconds for 2 simultaneous queries reinforce my guess that it is a database issue, not an Asterisk issue. What sort of response time to you get if you issue the queries from the mysql command prompt? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] DNS reload on trunks for outgoing calls
Is there any fix or workaround for the DNS problem (old standing bug that when the box starts and domain names do not resolve quickly enough from DNS then asterisk stops using the outgoing trunks. I read on the list before that it is considered a huge and unacceptable load for asterisk servers to try and resolve the domain names again after some time but it is rather annoying. I don't know about resources of other people but on my boxes i have some cpu cycles that could be used for that :) I now do nightly restarts of asterisk but it still means that at least for one day calls are flowing through expensive PSTN. If anybody knows of a workaround, would be most 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
[asterisk-users] Some minor configuration issues with queues
Hello list ! I have some configuration issues with queues, but I'm sure they are minor and for someone who has already configured queues it could be trivial. This is my queue configuration : [VC_support_queue] musicclass = default strategy = ringall timeout = 20 retry = 5 wrapuptime=15 autofill=yes autopause=no maxlen = 0 setinterfacevar=yes announce-frequency = 0 periodic-announce-frequency=0 announce-holdtime = no ; announce-round-seconds = 10 ; queue-thankyou= ;queue-youarenext = queue-youarenext ;queue-thereare = queue-thereare ;queue-callswaiting = queue-callswaiting ;queue-holdtime = queue-holdtime ;queue-minutes = queue-minutes ;queue-seconds = queue-seconds ;queue-thankyou = queue-thankyou ;queue-lessthan = queue-less-than ;queue-reporthold = queue-reporthold ;periodic-announce = queue-periodic-announce ; monitor-format = gsm|wav|wav49 ; monitor-type = MixMonitor joinempty = no ; leavewhenempty = yes eventwhencalled = no ; QueueMemberStatus eventmemberstatus = no reportholdtime = no ringinuse = no memberdelay = 0 ; timeoutrestart = no member = SIP/VCsupport,1,Jonas member = Agent/VCjoeri,2,Joeri First problem : Although there is nobody in the queue, the caller is still inserted into the queue : vps2301*CLI queue show VC_support_q has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2, SL:0.0% within 0s Members: Jonas (SIP/VCsupport) with penalty 1 (Unavailable) has taken no calls yet Joeri (Agent/VCjoeri) with penalty 2 (Unavailable) has taken no calls yet No Callers [Jan 4 09:34:40] -- Executing [...@center:2] Queue(IAX2/zoiper-5307, VC_support_queue|r) in new stack Second problem : When a caller calls in and there is someone available in the queue, there is no music-on-hold while the caller waits. When the caller is directed to the agent there should be a ringtone indicating that the agent is called, but it stays silent. No music on hold, no ringtone... (this is a SIP-channel) Thank you for pointing out the misconfiguration. Kind regards, Jonas. ___ -- 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] Skype for Asterisk
On 30 Dec 2009, at 19:43, vijay.go...@alliance-infotech.com wrote: Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine. case 2: This skype account (rexesbposolutions) has been assigned with a online virtual number (00 44 20 ). If somebody dial this number from their landline/cellphone, call is transfered to Asterisk queue but it shows some problem related to G729 codecs. following are Asterisk CLI log: Executing [...@skypeincoming:1] Answer(Skype/rexesbposolutions-084159e8, ) in new stack -- Executing [...@skypeincoming:2] Wait(Skype/rexesbposolutions-084159e8, 5) in new stack -- Executing [...@skypeincoming:3] GotoIfTime(Skype/rexesbposolutions-084159e8, 9:00-18:00|mon-fri|*|*?sky|s|1) in new stack -- Goto (sky,s,1) -- Executing [...@sky:1] Playback(Skype/rexesbposolutions-084159e8, enter) in new stack -- Skype/rexesbposolutions-084159e8 Playing 'enter' (language 'en') [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) [Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 4 -- Executing [...@sky:2] Queue(Skype/rexesbposolutions-084159e8, markq|t|||900) in new stack -- Started music on hold, class 'default', on Skype/rexesbposolutions-084159e8 [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin) [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:251 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/manolo_camp-morning_coffee': No such file or directory -- Stopped music on hold on Skype/rexesbposolutions-084159e8 [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:204 moh_files_release: Unable to restore channel 'Skype/rexesbposolutions-084159e8' to format '2' -- Playing periodic announcement [Dec 29 17:27:34] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) -- Skype/rexesbposolutions-084159e8 Playing 'queue' (language 'en') [Dec 29 17:27:43] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 2 == Spawn extension (sky, s, 2) exited non-zero on 'Skype/rexesbposolutions-084159e8' [Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call following are output of some commands:- *CLI core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- --- - gsm- -222 21 26 --2 - ulaw- 2-12 21 26 --2 - alaw- 21-2 21 26 --2 - g726aal2- 222- 21 26 --2 - adpcm- 2222 -1 26 --2 - slin- 1111 1- 15 --1 - lpc10- 2222 21 -6 --2 - g729- 6666 65 6- --6 - speex- ---- -- -- --- - ilbc- ---- -- -- --- - g726- 2222 21 26 --- - g722- ---- -- -- --- - *CLI help g729 g729 show hostid Show G.729 Host-ID g729 show licenses Show G.729 Licenses and Usage g729 show version Show G.729 Module Version *CLI g729 show hostid Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be *CLI g729 show licenses 0/0 encoders/decoders of 1 licensed channels are currently in use Licenses Found: File: ***-*.lic -- Key: ***-* -- Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be -- Channels: 1 (Expires: 2029-11-30) (OK) *CLI g729 show version Digium G.729A Module Version 1.4_3.1.4 (optimized for i686_32)
[asterisk-users] Dahdi and oslec
Looking at the source in the rpms from the asterisk package site appears that oslec is not built and enabled for the kmod rpms. Anyone know an existing repo or have direction on how to enable this to built for those rpms? Thanks, jlc ___ -- 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] Realtime Queue Members Not Ringing
Hi, So I'm using Asterisk Realtime Queues and Queue members on 1.4.28. I've noticed if there are no people in the queue when a call enters, even after a queue member enters, the call is never rang to him. From the debug, it seems that Asterisk is only grabbing the queue member list upon entering the queue. And not again until another call enters the queue. As a result, a caller will sit in the queue for an unknown amount of time until the the next caller enters the queue. My next thought was, 'well, I'll leave the queue member in there and just pause him when he goes on break'... but the same thing occurs. If the caller enters the queue when the member is paused, Asterisk continues to see him paused until another call enters the queue. So my question is this, is this fixed in 1.6.x? Or does anyone else even see this as a problem/bug? Might I suggest a fix of if the queue is empty upon entry of a call, Asterisk checks back every 15 seconds (or whatever) to refresh the queue member list. Btw, if a 'queue show queuename' is done from CLi, or QUEUE_MEMBER_COUNT() is used it will grab a new member list, and the call will ring through. Just seems like a bad behavior. Any thoughts? --Robert ___ -- 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] ZapRAS priviledge error
Hi, I'm trying to get ZapRAS working but not getting very far.. Asterisk CLI shows: WARNING[3355]: app_zapras.c:173 run_ras: wait4 returned -1: No child processes and /var/log/messages shows: using the plugin option requires root privilege Can anyone shed any light on this and any fix? Googling the error doesn't find much.. I'm not sure what 'plugin' it is talking about, I'm not passing any plugin option to zapras.. Ta, Will ___ -- 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] Inquiry:Asterisk different codec schemes?
hadi motamedi wrote: Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you No, you are not understanding the SDP offer/answer model properly. If one endpoint offers codecs A, B and C in its SDP, it is willing to *receive* media in those formats. The receiver of that offer can choose to send media to the offerer in any of those formats, at any time. If the answering endpoint includes only codec B in its SDP, then it is willing to *receive* only codec B. In that scenario, it is possible for media to flow from endpoint 1 to endpoint 2 using codec B, and from endpoint 2 to endpoint 1 using codec A (or C), but this will not happen if Asterisk is an endpoint in this scenario. When Asterisk receives a media frame, if the format of that frame is not the format that it is currently sending to the other endpoint, it will switch to that format automatically. If it cannot do so because the other endpoint did not offer to receive that format, then the call's audio will probably fail. This is the reason why I responded before that Asterisk does not support asymmetric formats in a media session. In reality, it is extremely uncommon for a SIP endpoint to want to send media in a format that it is not also willing to receive; in fact, I can't say I've ever seen this situation arise in any testing I've done or in any issues reported in our issue tracker. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] ZapRAS priviledge error
Will Payne wrote: I'm trying to get ZapRAS working but not getting very far.. Asterisk CLI shows: WARNING[3355]: app_zapras.c:173 run_ras: wait4 returned -1: No child processes and /var/log/messages shows: using the plugin option requires root privilege Can anyone shed any light on this and any fix? Googling the error doesn't find much.. I'm not sure what 'plugin' it is talking about, I'm not passing any plugin option to zapras.. ZapRAS forks off to pppd to handle the PPP session, it does not implement PPP itself. You will have to be running Asterisk as root for this to work, or provide a wrapper for pppd that ZapRAS can execute with the suid bit set so that pppd runs with root privileges. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] MYSQL queries from dial plan
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Neeraj Chand Sent: Monday, January 04, 2010 1:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] MYSQL queries from dial plan [mysql dialplan function] This currently takes about 4 seconds to complete. If I run two simultaneous queries, this goes up to about 9 seconds for both queries to complete. Is there a way that I can bring this time down? How long does the query take when executed at the MySQL command line? In my experience there is no perceptible Asterisk-related delay in executing MySQL from the dialplan versus ODBC versus the MySQL command line. Is DNS involved or do you access MySQL by IP? A long time ago we had a poorly written LCR routine that ran for about 3 seconds on a large set of tables. A little bit of intelligence and indexing has brought that down to 0.3 seconds on old single-core hardware. sl ___ -- 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] DNS reload on trunks for outgoing calls
On 4 Jan 2010, at 08:34, Remco Barendse wrote: Is there any fix or workaround for the DNS problem (old standing bug that when the box starts and domain names do not resolve quickly enough from DNS then asterisk stops using the outgoing trunks. I read on the list before that it is considered a huge and unacceptable load for asterisk servers to try and resolve the domain names again after some time but it is rather annoying. I don't know about resources of other people but on my boxes i have some cpu cycles that could be used for that :) I now do nightly restarts of asterisk but it still means that at least for one day calls are flowing through expensive PSTN. If anybody knows of a workaround, would be most welcome Install a resolver locally. ___ -- 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] caller getting cut off intermittently
I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x) Everything works fine (incoming/outgoing audio etc.) except occasionally an incoming caller is cut off whilst the called extension stays in the call and can hear a DTMF tone (multimon recognises it as tone D). The asterisk log file shows the call stays active despite the incoming caller being cut off. This has happened to all our extensions at some point (a combination of Snoms and Funkwerks). It happens fairly infrequently, and can happen at any point during a call. The public Lenny server's asterisk config is exactly the same as our LAN Ubuntu asterisk server where we never had this problem. The only difference is that the ITSP trunk is now ulaw rather than ilbc. Can anyone help? Relevant files below (trunk and extension codecs are both ulaw) John example extension in sip.conf: [203] type=friend username=203 secret=xx host=dynamic dtmfmode=inband call-limit=2 qualify=yes nat=yes /var/log/asterisk/messages: [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00, 0?bankhols|200|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00, 08:30-18:00|mon-fri|*|*?day|100|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Goto (day,100,1) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Launched AGI Script /home/john/phpagi/lookup [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- AGI Script /home/john/phpagi/lookup completed, returning 0 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:1] Set(SIP/301x-09f74a00, CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:2] Monitor(SIP/301x-09f74a00, wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:4] Dial(SIP/301x-09f74a00, SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 203 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 206 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 207 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 220 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/206-0a005eb8 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/207-09fe2c98 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:59] VERBOSE[10712] logger.c: -- SIP/203-0a001138 answered SIP/301x-09f74a00 ___ -- 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] Multiple Digium cards with one NFAS trunkgroup
From reading the documentation that came with dahdi-tools I gathered that the second span should be span=2,2... designating it as a backup timing source in case your primary span=1,1... should die. I am not sure that you can designate two primary timing sources and have seamless failover in a situation like the one you described. ___ -- 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] ZapRAS priviledge error
On 4 Jan 2010, at 13:48, Kevin P. Fleming wrote: ZapRAS forks off to pppd to handle the PPP session, it does not implement PPP itself. You will have to be running Asterisk as root for this to work, or provide a wrapper for pppd that ZapRAS can execute with the suid bit set so that pppd runs with root privileges. OK.. I'd tried to setuid pppd so had wondered if it was just permissions. Now have pppd apparently starting correctly but am now getting: Jan 4 16:07:15 asterisk pppd[29485]: Plugin zaptel.so loaded. Jan 4 16:07:15 asterisk pppd[29485]: Zaptel Plugin Initialized Jan 4 16:07:15 asterisk pppd[29485]: Using zaptel device 'stdin' Jan 4 16:07:15 asterisk pppd[29485]: pppd 2.4.4 started by root, uid 0 Jan 4 16:07:15 asterisk pppd[29485]: Zaptel device is 'stdin' Jan 4 16:07:15 asterisk pppd[29485]: Device 'stdin' does not appear to be a zaptel device Jan 4 16:07:15 asterisk pppd[29485]: Exit. ..which is probably just because I'm doing something stupid (I've never touched PPP on linux and am still figuring asterisk out..) I'd tried initiating a call using: originate ZAP/g0d/0xx application ZapRas xx|xx|xx... Will this work? I'm looking to periodically nudge Asterisk into making an ISDN connection, setting up PPP and then (possibly by then starting an AGI script) grabbing a file by FTP over the PPP link. If I'm overcomplicating it or going about it completely the wrong way, a point in the right direction would be nice :) Will___ -- 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] ZapRAS priviledge error
Will Payne wrote: I'm looking to periodically nudge Asterisk into making an ISDN connection, setting up PPP and then (possibly by then starting an AGI script) grabbing a file by FTP over the PPP link. If I'm overcomplicating it or going about it completely the wrong way, a point in the right direction would be nice :) It is doubtful you'll be able to accomplish that, certainly not without some seriously ugly hacking. First off, I don't think that PPPD will even be invoked with the proper arguments for it to be the 'client' end of the connection, but even if it is, the Asterisk dialplan will halt execution until PPPD returns, so there's no way you are going to be able to execute an AGI or System() or anything to take actions over the PPP link. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] ZapRAS priviledge error
On 4 Jan 2010, at 16:28, Kevin P. Fleming wrote: Will Payne wrote: I'm looking to periodically nudge Asterisk into making an ISDN connection, setting up PPP and then (possibly by then starting an AGI script) grabbing a file by FTP over the PPP link. If I'm overcomplicating it or going about it completely the wrong way, a point in the right direction would be nice :) It is doubtful you'll be able to accomplish that, certainly not without some seriously ugly hacking. First off, I don't think that PPPD will even be invoked with the proper arguments for it to be the 'client' end of the connection, but even if it is, the Asterisk dialplan will halt execution until PPPD returns, so there's no way you are going to be able to execute an AGI or System() or anything to take actions over the PPP link. Unfortunately, an ugly hack might have to do.. Params - should be able to work around this, even if I have to use a wrapper. PPPD halting the dialplan - I'll fork off a different process to watch for a connection and make the transfer. I can just tell pppd to connect for a minimum of 'x' seconds and then let Asterisk hang up. .. which still leaves me in the same position of wondering why I'm getting this Device 'stdin' does not appear to be a zaptel device error... Will ___ -- 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] Script to show asterisk stuff
Hello folks. I'm looking into having a web page displaying asterisk callers. We are a call centre, and having operators answering calls at home or whatever, they would need to have a real time application to display how manny callers are queuing, for how long etc. At first, I thought of phpagi. It connects to the manager and does a core show channels concise. This has most of the info I want. After tweaking with php to parse the text to exatcly how I wanted, I found out that the script would be slow if it was self refreshing (say 2 secs) and with about 30 people opening it at the same time. So now I was thinking in a script that would connect to the Manager, and parse that output into a mysql table. A Web page would consult the mysql table, showing the wanted results. Then I thought twice and maybe some of you already developed a situation like this and would not mind sharing? I don't mind sharing the little I done so far, if anyone is interested. Thanks all ___ -- 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] DNS reload on trunks for outgoing calls
Put the commonly used domain names + appropriate ips into /etc/hosts? John 2010/1/4 Steve Howes steve-li...@geekinter.net: On 4 Jan 2010, at 08:34, Remco Barendse wrote: Is there any fix or workaround for the DNS problem (old standing bug that when the box starts and domain names do not resolve quickly enough from DNS then asterisk stops using the outgoing trunks. I read on the list before that it is considered a huge and unacceptable load for asterisk servers to try and resolve the domain names again after some time but it is rather annoying. I don't know about resources of other people but on my boxes i have some cpu cycles that could be used for that :) I now do nightly restarts of asterisk but it still means that at least for one day calls are flowing through expensive PSTN. If anybody knows of a workaround, would be most welcome Install a resolver locally. ___ -- 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] Script to show asterisk stuff
On 4 Jan 2010, at 16:46, Tiago Geada wrote: Hello folks. I'm looking into having a web page displaying asterisk callers. We are a call centre, and having operators answering calls at home or whatever, they would need to have a real time application to display how manny callers are queuing, for how long etc. At first, I thought of phpagi. It connects to the manager and does a core show channels concise. This has most of the info I want. After tweaking with php to parse the text to exatcly how I wanted, I found out that the script would be slow if it was self refreshing (say 2 secs) and with about 30 people opening it at the same time. So now I was thinking in a script that would connect to the Manager, and parse that output into a mysql table. A Web page would consult the mysql table, showing the wanted results. Or, if you want less work.. have a script which connects to the manager, formats the data and creates an HTML page. Then wait x seconds and loop. Then, home workers just view that one static page and use a meta-refresh or something.. Only one script is doing any real work and serving a static page to clients shouldn't overload the server. Will ___ -- 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] Script to show asterisk stuff
2010/1/4 Will Payne w...@teambadger.co.uk On 4 Jan 2010, at 16:46, Tiago Geada wrote: Hello folks. I'm looking into having a web page displaying asterisk callers. We are a call centre, and having operators answering calls at home or whatever, they would need to have a real time application to display how manny callers are queuing, for how long etc. At first, I thought of phpagi. It connects to the manager and does a core show channels concise. This has most of the info I want. After tweaking with php to parse the text to exatcly how I wanted, I found out that the script would be slow if it was self refreshing (say 2 secs) and with about 30 people opening it at the same time. So now I was thinking in a script that would connect to the Manager, and parse that output into a mysql table. A Web page would consult the mysql table, showing the wanted results. Or, if you want less work.. have a script which connects to the manager, formats the data and creates an HTML page. Then wait x seconds and loop. Then, home workers just view that one static page and use a meta-refresh or something.. Only one script is doing any real work and serving a static page to clients shouldn't overload the server. Will __ Hi Will. Thanks for replying. That was sort of my second thought. But once I connect to the manager I can listen to all the events, Calls comming in, which extension they are dialed to, lots of info... so I just got sort of confused for whitch path I should take. I guess I will do just that. Thanks _ -- 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] H323 Disconnects after 15+ minutes
I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem: I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected. From what I can tell, the cause is condition 20 on ooh323. Any suggestions as to the cause? http://www.elastix.org/component/option,com_fireboard/Itemid,55/func,view/catid,3/id,41480/lang,en/#42715 Dec 29 10:25:01 VERBOSE [15027] logger.c: -- Remote UNIX connection Dec 29 10:25:01 VERBOSE [31438] logger.c: -- Remote UNIX connection disconnected Dec 29 10:26:01 WARNING [31413] chan_ooh323.c: Don't know how to indicate condition 20 on ooh323c_9 Dec 29 14:42:06 VERBOSE [349] logger.c: -- SIP/5034-1b1aa680 is ringing Dec 29 14:42:09 VERBOSE [349] logger.c: -- SIP/5034-1b1aa680 answered OOH323/denver-eaf3 Dec 29 14:42:09 WARNING [349] chan_ooh323.c: Don't know how to indicate condition 20 on ooh323c_18 Dec 29 15:02:55 VERBOSE [410] logger.c: -- Remote UNIX connection disconnected Dec 29 15:04:01 WARNING [349] chan_ooh323.c: Don't know how to indicate condition 20 on ooh323c_18 Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-dial:1] Macro(OOH323/denver-eaf3, hangupcall) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:1] ResetCDR(OOH323/denver-eaf3, w) in new stack Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: ResetCDR Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:2] NoCDR(OOH323/denver-eaf3, ) in new stack Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: NoCDR Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:3] GotoIf(OOH323/denver-eaf3, 1?skiprg) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: -- Goto (macro-hangupcall,s,6) Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: GotoIf Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:6] GotoIf(OOH323/denver-eaf3, 1?skipblkvm) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: -- Goto (macro-hangupcall,s,9) Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: GotoIf Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:9] GotoIf(OOH323/denver-eaf3, 1?theend) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: -- Goto (macro-hangupcall,s,11) Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: GotoIf Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:11] Hangup(OOH323/denver-eaf3, ) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'OOH323/denver-eaf3' in macro 'hangupcall' Dec 29 15:04:01 VERBOSE [349] logger.c: == Spawn h extension (macro-dial, h, 1) exited non-zero on 'OOH323/denver-eaf3' ___ -- 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] Register sip FXO per gateway
How to register/configure Sip accounts to register per gateway? All the accounts I have are registered individually but with mix (FXO/FXS) AudioCodes MP-114 this does not work, I have to have FXO port registered per gateway (not individually). -- Joseph ___ -- 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] SIP Listen Multiple Ports
Hello, I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Olle E. Johansson wrote: No, Asterisk only supports one port. You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? If so, what would I be losing in not letting OpenSER do it? I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). Thanks, Vikram. ___ -- 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] SIP Listen Multiple Ports
1 jan 2010 kl. 20.04 skrev Shariq Khan: I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). On Mon, 4 Jan 2010, Vikram Ragukumar wrote: Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? Yes, but the number of SIP control messages are usually insignificant compared to all the RTP packets. If so, what would I be losing in not letting OpenSER do it? All the goodness that OpenSER brings to the table. I just scratched the surface above with all the features of OpenSER/Kamailio/OpenSIPS. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] SIP Listen Multiple Ports
Could you explain this one a bit more... You run openSER on the same box as asterisk, and have multiple such boxes, with the purpose of failover? But if a box goes down with openser on it, then there is no forwarding. (And most phones can only reg with peer). If you move openSER to another independent box, then you have a single point of failure. I suspect I'm not understanding something correctly -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, January 04, 2010 2:34 PM To: Asterisk Users List Cc: asterisk@sedwards.com Subject: Re: [asterisk-users] SIP Listen Multiple Ports 1 jan 2010 kl. 20.04 skrev Shariq Khan: I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). On Mon, 4 Jan 2010, Vikram Ragukumar wrote: Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? Yes, but the number of SIP control messages are usually insignificant compared to all the RTP packets. If so, what would I be losing in not letting OpenSER do it? All the goodness that OpenSER brings to the table. I just scratched the surface above with all the features of OpenSER/Kamailio/OpenSIPS. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Dialout from Meetme conference
I am implementing one dialer type of application. In which i am first dialing one source number and sending it to conference and then starting dialing the different destination numbers. i have used meetme application of asterisk for this as i dont want to disconnect the main source number. Both numbers are being originated using AMI and in the context i have used MeetMe with the specified room. So, as soon as they got answered, they are being sent into the conference and they can hear each other and talk. Everything goes fine with the above mentioned settings. But, what i am trying to achieve is, I want to dial the other number from the meetme, so the source number can also hear the ringing sound of the other phone. So, the call must be originated from conference and get into conference as soon as the channel is originated. For this i have used channel redirect function of phpagi-asmanager.php to send the dialing channel of destination number to conference. By doing this i can hear the ring sound, but when the call is being answered, the first user of conference is not able to hear the voice of second user. I checked the asterisk CLI and debug the issue, And i found that the channel which we sent into conference without answer is having state Down. Can this be the issue of voice ? Sometime during the test i get it working but on the next test it fails. So i think this might be a bug in the asterisk source code Please suggest any method or patch to overcome this Thanks Shrikant Soni __ Information from ESET Smart Security, version of virus signature database 4739 (20100103) __ The message was checked by ESET Smart Security. http://www.eset.com ___ -- 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] SIP Listen Multiple Ports
Un-top-posting... On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). On Mon, 4 Jan 2010, Michelle Dupuis wrote: Could you explain this one a bit more... You run openSER on the same box as asterisk, and have multiple such boxes, with the purpose of failover? But if a box goes down with openser on it, then there is no forwarding. (And most phones can only reg with peer). If you move openSER to another independent box, then you have a single point of failure. I suspect I'm not understanding something correctly This is for a system advertised with those cheesy late night cable TV ads -- hot girl enticing you to call for a free chat and talk with all your friends... The hosts are located in a rural telco. The telco switch (Taqua 7000 as I remember) can hand off the call to a couple of IP addresses. Each of these addresses is on a separate host with OpenSER (it's been a few years) listening to 5060. Each of these hosts is also running Asterisk (1.2) listening on port 5061 There are no phones registering with any host. All calls come in through the Taqua or IAX when I need to test something. Each instance of OpenSER is configured (dispatcher.list) to distribute (no active load balancing) calls to all of the instances of Asterisk, including the instance running on the same host. If I want to take an instance of Asterisk down for maintenance, I just comment out the address associated with that instance out of dispatcher.list, restart the instances of OpenSER and wait for the in-progress calls to time out. If a host crashes, the Taqua detects that and doesn't send calls to that instance of OpenSER anymore. Each remaining instance of OpenSER will send calls to the remaining instances of Asterisk. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Realtime Queue Members Not Ringing
Robert Broyles wrote: Hi, So I'm using Asterisk Realtime Queues and Queue members on 1.4.28. I've noticed if there are no people in the queue when a call enters, even after a queue member enters, the call is never rang to him. From the debug, it seems that Asterisk is only grabbing the queue member list upon entering the queue. And not again until another call enters the queue. As a result, a caller will sit in the queue for an unknown amount of time until the the next caller enters the queue. My next thought was, 'well, I'll leave the queue member in there and just pause him when he goes on break'... but the same thing occurs. If the caller enters the queue when the member is paused, Asterisk continues to see him paused until another call enters the queue. So my question is this, is this fixed in 1.6.x? Or does anyone else even see this as a problem/bug? Might I suggest a fix of if the queue is empty upon entry of a call, Asterisk checks back every 15 seconds (or whatever) to refresh the queue member list. Btw, if a 'queue show queuename' is done from CLi, or QUEUE_MEMBER_COUNT() is used it will grab a new member list, and the call will ring through. Just seems like a bad behavior. Any thoughts? --Robert So 1.6.x doesn't have any improvement in this regard. Can someone point me in the right direction here? Maybe modify app_queue to refresh the queue member list from time to time? Or even after the periodic announcement would be good - anything would be an improvement. Right now I have a cron running every minute to 'queue show queuename' - but that's not scalable, especially after you have several queues you need to do this with. What's the logic in the current system? ___ -- 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] SIP Incoming / Inbound not working for Broadvoice (Asterisk PBX 1.6.1.6)
Thank you Doug and Nguyen. I have had your recommendation but I still can
not get SIP inbound for my broadvoice line to work. My configuration and SIP
debugs are attached and just to recap I have done the following:
My outgoing works great. I can dial from one SIP extension to another
internally with no problem. The issue is incoming SIp is not working and
goes directly ro Voice mail.
1) UDP port 5060, 691 10K-20K are allowed through the Firewall (Port forward
to Asterisk server)
2) I have 2 SIP extensions 5000 and 5002
3) Made sure I have the correct context on the incoming extension
4) Added nat=yes and qualify=yes under SIP.conf on both extensions
If anyone knows if there are any other ideas or what debug to turn on pls
let me know. I would appreciate if anyone from Broadvoice Support or
Broadvoice user can give pointers as well.
Thanks.
On Wed, Dec 30, 2009 at 10:57 PM, Doug d...@natel.net wrote:
At 18:22 12/30/2009, Qurba Joog wrote:
You are correct.. I had the correct context on my current production
configuration I just copied from an older saved file.. So the
[enterbroadvoice] has a context of incoming and incoming is defined in the
extensions.con. But still have the same problem with incoming jumping
directly into Broadvoice VM. I even changed it to IAX2 extension.
Any help would be appreciated
I don't this for any of your peers:
nat=yes
On Fri, Dec 25, 2009 at 10:43 AM, Qurba Joog mailto:qurbaj...@gmail.com
qurbaj...@gmail.com wrote:
Thanks very much for your reply Nguyen. I read and re-read the url you
sent.. Sorry I'm new to this but are you saying the [incoming] context is
wrong? Please indicate what is wrong.
If you look at the SIP.conf [enter_broadvoice] I have [incoming] context
defined and that is the one I have called out in the extensions.conf
Thanks.
On Fri, Dec 25, 2009 at 5:20 AM, Nguyen Quang Tri http://kihote.am
kihote.am@http://gmail.comgmail.com wrote:
Wrong context for incoming,
you can read
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
and
http://www.voip-info.org/wiki/view/IPKall
http://www.voip-info.org/wiki/view/IPKall
2009/12/25 --[ UxBoD ]-- mailto:ux...@splatnix.netux...@splatnix.net
- Qurba Joog mailto:qurbaj...@gmail.comqurbaj...@gmail.com
wrote:
| Hello,
|
| Please forgive me if I'm repeating this post. I have searched and looked
for similar problem with a solution but have not see a similar one.
|
| My outgoing SIP and other channels work fine but the incoming/inbound
SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is
registered when I look at the SIP registry. I have turned on SIP Debug and
it is below.
|
| Anyone know why even when SIP has registered I do not see incoming
calls?
|
| Thanks,
|
|
| --extensions.conf
| [global]
|
| [general]
| bindport=5060
| bindaddr = 0.0.0.0
| deny=http://0.0.0.0/0.0.0.0MailScanner has detected a possible fraud
attempt from 0.0.0.0 claiming to be MailScanner warning: numerical links
are often malicious:http://0.0.0.0/0.0.0.0 0.0.0.0/0.0.0.0
| externhost=http://xyz.dyndns.orgxyz.dyndns.org
| localnet = http://192.168.1.0/255.255.255.0MailScanner has detected a
possible fraud attempt from 192.168.1.0 claiming to be MailScanner
warning: numerical links are often malicious:
http://192.168.1.0/255.255.255.0 192.168.1.0/255.255.255.0
| disallow=all
| allow=ulaw
| allow=gsm
| delayreject=yes
| nochecksums=no
| allowguest=no
| delayreject=yes
| pedantic=no
|
| register = 703xxxy...@sip.broadvoice.com:s
http://ecurepassword:703xxxy...@sip.broadvoice.com/5000
ecurepassword:703xxxy...@sip.broadvoice.com/5000
|
| [5000]
| type=friend
| context=internal-phones
| secret=xxx
| qualify=yes
| host=dynamic ; behind nat
| dtmfmode=rfc2833
|
| [5002]
| type=friend
| context=internal-phones
| secret=test
| qualify=yes
| host=dynamic ; behind nat
| nat=yes
| dtmfmode=rfc2833
|
| [enter_broadvoice]
| type=peer
| user=phone
| host=http://sip.broadvoice.comsip.broadvoice.com
| fromdomain=http://sip.broadvoice.comsip.broadvoice.com
| fromuser=703XXX
| secret=securepassword
| username=703XXX
| insecure=very
| ;insecure=port,invite
| context=incoming
| authname=703XXX
| dtmfmode=inband
| dtmf=inband
| ;Disable canreinvite if you are behind a NAT
|
| canreinvite=no
|
| extensions.conf
|
| [globals]
|
| [general]
|
| autofallthrough=yes
|
|
| [incoming_calls]
|
| exten = 1703XXX,1,Dial(SIP/5000)
|
| [internal-phones]
|
| include = outgoing
| exten = 5000,1,Dial(SIP/5000,20)
| exten = 5002,1,Dial(SIP/5002,20)
|
|
| [outgoing]
|
| exten = _X.,1,NoOp()
| exten = _X.,n,Dial(SIP/enter_broadvoice/${EXTEN})
|
| SIP Registry--
| -*CLI sip show registry
| Host dnsmgr Username Refresh State
[asterisk-users] lpc10
Is there a way to not compile in lpc10 support using the ./configure command? ./configure --disable-lpc10 or something like that? if that is not available whats the easiest way to remove lpc10 support with out doing the make menuselect. I want to do it automatically at install not have to enter a command goto codecs and unselect lpc10. Thanks, Jerry ___ -- 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] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE
Hi guys, Am having a strange SIP problem in my call centre. The call centre has about 70 SIP agents (some of the are using SIP hard phones, other SIP softphones), and occasionally most of the SIP peers (hardphones and softphones) become UNREACHABLE and then after few second again REACHABLE. Some hardphones and softphones work perfectly normal during that period (even normally responding to OPTIONS message), but most of them get UNREACHABLE. I don't have NAT - phones and Asterisk are in the same subnet, so nothing complicated really (regarding network configuration). I'm currently suspecting my network to be the problem, but I would just like to confirm with you guys, if you have any similar experiences, what could be causing this? Please, see bellow one of the sample SIP traces. Regards, Alex Jan 1 11:17:42 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060: OPTIONS sip:testpho...@165.11.1.41 SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport From: asterisk sip:aster...@165.11.1.50;tag=as02e1afaa To: sip:testpho...@165.11.1.41 Contact: sip:aster...@165.11.1.50 Call-ID: 3fa169320586bad01cd93bd87adf1...@165.11.1.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:45 VERBOSE[6046] logger.c: Retransmitting #1 (no NAT) to 165.11.1.41:5060: OPTIONS sip:testpho...@165.11.1.41 SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport From: asterisk sip:aster...@165.11.1.50;tag=as02e1afaa To: sip:testpho...@165.11.1.41 Contact: sip:aster...@165.11.1.50 Call-ID: 3fa169320586bad01cd93bd87adf1...@165.11.1.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:46 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now UNREACHABLE! Last qualify: 14 Jan 1 11:17:56 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060: OPTIONS sip:testpho...@165.11.1.41 SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport From: asterisk sip:aster...@165.11.1.50;tag=as796f6356 To: sip:testpho...@165.11.1.41 Contact: sip:aster...@165.11.1.50 Call-ID: 3367c4dc6cbdd57d67b0c5b53d549...@165.11.1.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:56 VERBOSE[6046] logger.c: -- SIP read from 165.11.1.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport From: asterisk sip:aster...@165.11.1.50;tag=as796f6356 To: sip:testpho...@165.11.1.41;tag=5A4BF5F8-460290A9 CSeq: 102 OPTIONS Call-ID: 3367c4dc6cbdd57d67b0c5b53d549...@165.11.1.50 Contact: sip:testpho...@165.11.1.41 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.0.0047 Content-Length: 0 Jan 1 11:17:56 VERBOSE[6046] logger.c: --- (10 headers 0 lines) --- Jan 1 11:17:56 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now REACHABLE! (16ms / 1ms) ___ -- 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] SIP Listen Multiple Ports
Hello, 1 jan 2010 kl. 20.04 skrev Shariq Khan: I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. I like to configure systems with OpenSER running on each box, forwarding calls to Asterisk across the same set of boxes for redundancy, load balancing, and maintenance (being able to take an instance of Asterisk or an entire box out of production). On Mon, 4 Jan 2010, Vikram Ragukumar wrote: Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? Yes, but the number of SIP control messages are usually insignificant compared to all the RTP packets. If so, what would I be losing in not letting OpenSER do it? All the goodness that OpenSER brings to the table. I just scratched the surface above with all the features of OpenSER/Kamailio/OpenSIPS. My customer is keen on using a hardware bridge to maximize throughput and also allow multiple servers. My boss is pressing me to maintain Kamailio and rtpproxy compatibility, and understand the tradeoffs in satisfying both. The link below has a diagram showing the way I'm going now: http://signalogic.com/images/openser_asterisk_sysconfig_dataflow.jpg Will the bridge preclude me from gracefully modifying my code to use more Kamailio functionality, if needed? Thanks and Regards, Vikram. ___ -- 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] cheap ip phone with auto-answer
On 29/12/09 10:22 AM, Leif Neland wrote: I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Use some of the Chinese PA1688 or AR1688 phones - support auto answer, IAX/SIP etc. Prices around $45 -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) ___ -- 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] T.38 ITSP?
Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send (because they use fax-to-email for receive), It should be trivial to relay via T-38 to any of our asterisk servers that DO have a PRI or copper loop, and send (or queue-up) the fax to be sent. It's the 'local number' for 'traditional receive' that looking to be the harder problem. Is anybody using an ITSP for inbound T-38 fax with 'local' numbers? Thanks! -Karl ___ -- 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] SIP Listen Multiple Ports
On Sun, 3 Jan 2010, Steve Edwards wrote: You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple addresses and ports and forward to Asterisk on the same or different boxes. On Mon, 4 Jan 2010, Vikram Ragukumar wrote: Would it be more efficient to use libnetfilter_queue() to listen to specific addresses / ports and forward to Asterisk? Yes, but the number of SIP control messages are usually insignificant compared to all the RTP packets. On Mon, 4 Jan 2010, Vikram Ragukumar wrote: My customer is keen on using a hardware bridge to maximize throughput and also allow multiple servers. My boss is pressing me to maintain Kamailio and rtpproxy compatibility, and understand the tradeoffs in satisfying both. The link below has a diagram showing the way I'm going now: http://signalogic.com/images/openser_asterisk_sysconfig_dataflow.jpg Will the bridge preclude me from gracefully modifying my code to use more Kamailio functionality, if needed? I won't claim to have any insight into your system, but... Your diagram shows all SIP messages (unencrypted and decrypted) flowing through Kamailio. My guess is that you would have access to all Kamailio features. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] AGI and embargeability
Hi, This is a naive question, but is there a way in my AGI script to simultaneously play audio and listen for DTMF or voice responses? I've heard VOIP hackers call this inbargeability; it's the ability to barge in to a playing audio clip. I'm planning to use Lumenvox for the DTMF and voice recognition, BTW. Not sure if that matters. Many thanks to anyone who can lend me a clue about this, -- Quinn Weaver Consulting, LLC Full-stack web design and development http://quinnweaver.com/ 510-520-5217 ___ -- 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] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE
Have you tried something like qualify=10 ? ___ -- 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] AGI and embargeability
Sure, as long as you use whatever is equivalent to the Background() dial plan app, or Background() itself. On 01/04/2010 08:41 PM, Quinn Weaver wrote: Hi, This is a naive question, but is there a way in my AGI script to simultaneously play audio and listen for DTMF or voice responses? I've heard VOIP hackers call this inbargeability; it's the ability to barge in to a playing audio clip. I'm planning to use Lumenvox for the DTMF and voice recognition, BTW. Not sure if that matters. Many thanks to anyone who can lend me a clue about this, -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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] AGI and embargeability
On Mon, 4 Jan 2010, Quinn Weaver wrote: This is a naive question, but is there a way in my AGI script to simultaneously play audio and listen for DTMF or voice responses? t2:vtpv:18:04:59 show agi stream file Usage: STREAM FILE filename escape digits [sample offset] Send the given file, allowing playback to be interrupted by the given digits, if any. Use double quotes for the digits if you wish none to be permitted. If sample offset is provided then the audio will seek to sample offset before play starts. Returns 0 if playback completes without a digit being pressed, or the ASCII numerical value of the digit if one was pressed, or -1 on error or if the channel was disconnected. Remember, the file extension must not be included in the filename. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] MYSQL queries from dial plan
On Jan 4, 2010, at 3:06 PM, Steve Edwards wrote: On Mon, 4 Jan 2010, Neeraj Chand wrote: I currently run small scale mysql queries from the dialplan [snip] This currently takes about 4 seconds to complete. If I run two simultaneous queries, this goes up to about 9 seconds for both queries to complete. 4 seconds for a query is a bit extreme. 9 seconds for 2 simultaneous queries reinforce my guess that it is a database issue, not an Asterisk issue. What sort of response time to you get if you issue the queries from the mysql command prompt? This have happened to me in various situations and is usually a result of reverse DNS lookups failing on the MySQL side. See: http://dev.mysql.com/doc/refman/5.0/en/dns.html for a solution to that if that is the issue. Best regards, Peter Lindqvist Voxion Ltd. www.voxion.net ___ -- 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] Inquiry:Asterisk different codec schemes?
On Mon, Jan 4, 2010 at 1:46 PM, Kevin P. Fleming kpflem...@digium.comwrote: hadi motamedi wrote: Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you No, you are not understanding the SDP offer/answer model properly. If one endpoint offers codecs A, B and C in its SDP, it is willing to *receive* media in those formats. The receiver of that offer can choose to send media to the offerer in any of those formats, at any time. If the answering endpoint includes only codec B in its SDP, then it is willing to *receive* only codec B. In that scenario, it is possible for media to flow from endpoint 1 to endpoint 2 using codec B, and from endpoint 2 to endpoint 1 using codec A (or C), but this will not happen if Asterisk is an endpoint in this scenario. When Asterisk receives a media frame, if the format of that frame is not the format that it is currently sending to the other endpoint, it will switch to that format automatically. If it cannot do so because the other endpoint did not offer to receive that format, then the call's audio will probably fail. This is the reason why I responded before that Asterisk does not support asymmetric formats in a media session. In reality, it is extremely uncommon for a SIP endpoint to want to send media in a format that it is not also willing to receive; in fact, I can't say I've ever seen this situation arise in any testing I've done or in any issues reported in our issue tracker. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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 Thank you very much for correcting me . ___ -- 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] Inquiry:Asterisk sending dialed digits in one-by-one digit format?
Dear All Further to my previous inquiry regarding Asterisk sending dialed digits in one-by-one digit format when we had ISDN PRI link with the PSTN switch , you told me that we are expected to enable overlap dialing . At now , we have the same configuration but sip connection to the external sip server . Please be informed that the sip inbound outbound is working correctly but we are expected to send the dialed digits in one-by-one digit format . Can you please let me know what is applicable here in our case ? Thank you ___ -- 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] automatic dial from database
Hi , Can any one tell me that how to automatically dial a list of numbers from database .I have seen a methodology in the post but am not clear vth that. Thanks Pinky ___ -- 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
