Re: [asterisk-users] multiple call
Use an existing dialer like ViciDialer? l. 2009/10/14 kaustuva...@bbsr.syscomes.com Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager Thanks and regards -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] multiple call
On 15/10/09 4:42 AM, Faheem wrote: Through Asterisk AMI, you can not dial multiple number at the same time. If you are going to implement a concurrent call scenario, then AMI would not be a valid choice. Multiple calls can be implemented with callfile. Totally incorrect. We do hundreds of simultaneous calls at the same time using the Asterisk Manager. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Calls hang up after 20 seconds
Hello. I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason. The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and one of them answer, the call ends itself after 20 seconds from the answer. I've tried many configuration in sip.conf, but no one solved the problem. Log from /var/log/asterisk/messages: [Oct 8 15:49:05] WARNING[10659] chan_sip.c: Hanging up call e3f0204b-b35811de-bba58889-6d53c...@83.211.2.220 - no reply to our critical packet. and from CLI: [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 59e62874-b35911de-9a598915-c5a6b...@195.62.226.16 for seqno 101 (Critical Response) [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1972 retrans_pkt: Hanging up call 59e62874-b35911de-9a598915-c5a6b...@195.62.226.16 - no reply to our critical packet. == Spawn extension (incoming, 03411885583, 4) exited non-zero on 'SIP/03411885583-081e0d78' (peer 101 was not connected at this time, but Asterisk also hags up with all the peers connected) Any idea? Thanks in advance. -- Gianni Fioretta - gianni.fiore...@yetopen.it YetOpen S.r.l. - http://www.yetopen.it/ Via Previati 72 - 23900 Lecco - ITALY - Tel 0341 220 205 - Fax 178 607 8199 D.Lgs. 196/2003 Si avverte che tutte le informazioni contenute in questo messaggio sono riservate ed a uso esclusivo del destinatario. Nel caso in cui questo messaggio Le fosse pervenuto per errore, La invitiamo ad eliminarlo senza copiarlo, a non inoltrarlo a terzi e ad avvertirci non appena possibile. Grazie. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Calls hang up after 20 seconds
Hi We've had this a few times and never got to the bottom of exactly why it happens but stopped it by upgrading the phone and the router it was going through to the latest firmware versions. Hope that helps Ish Gianni Fioretta wrote: Hello. I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason. The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and one of them answer, the call ends itself after 20 seconds from the answer. I've tried many configuration in sip.conf, but no one solved the problem. Log from /var/log/asterisk/messages: [Oct 8 15:49:05] WARNING[10659] chan_sip.c: Hanging up call e3f0204b-b35811de-bba58889-6d53c...@83.211.2.220 - no reply to our critical packet. and from CLI: [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 59e62874-b35911de-9a598915-c5a6b...@195.62.226.16 for seqno 101 (Critical Response) [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1972 retrans_pkt: Hanging up call 59e62874-b35911de-9a598915-c5a6b...@195.62.226.16 - no reply to our critical packet. == Spawn extension (incoming, 03411885583, 4) exited non-zero on 'SIP/03411885583-081e0d78' (peer 101 was not connected at this time, but Asterisk also hags up with all the peers connected) Any idea? Thanks in advance. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Queues with unavailable members
Elliot Otchet elliot.otc...@callingcircles.com writes: Have you tried autopause=yes in your queue configuration? You can then unpause the member by either the dialplan (e.g. having the cell phone user log back in) or using an AMI based program to change the paused state. You can read more about the latter here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html That looks very interesting, thank you! First of all though I need to avoid having them autopause just because they don't answer their phone. It should only happen if the call to their phone fails completely. I guess that could be done by not doing autopause but instead pausing manually in the context that the Local call passes through. That would also solve my second problem, which is that I need to pause it in all queues, not just one queue. The last challenge is to somehow unpause them after a while. In traditional programming that would be something like keeping a list of timeout,queuemember ordered by timeout, and then when every call comes in unpause and remove the ones where timeout expired... I'm not sure that I can make an ordered list in the dialplan though. I may have to resort to AGI, but I still need somewhere to actually store the list. Tricky. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Queues with unavailable members
Lenz Emilitri lenz.lo...@gmail.com writes: You could configure them as agents and have them log off automatically after a while they're not responding. Agents have to log in and wait for calls though, don't they? There used to be AgentCallbackLogin, but that has been replaced by dialplan code and chan_local. Otherwise a nice idea though. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Calls hang up after 20 seconds
--- On Thu, 10/15/09, Gianni Fioretta gianni.fiore...@yetopen.it wrote: I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason. The call comes from a SIP server Hi, This may or may not apply to your case: https://issues.asterisk.org/view.php?id=14652 Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk with a Cisco AS5300 gateway
Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec preference 2 g729r8 codec preference 3 g723r63 codec preference 4 g711ulaw ! voice class codec 500 codec preference 1 g729r8 codec preference 2 g723r63 ! controller E1 0 framing NO-CRC4 pri-group timeslots 1-31 description E1 Beta-Test interface Serial0 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial1 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial2 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial3 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial0:15 no ip address encapsulation ppp isdn switch-type primary-net5 no cdp enable voice-port 0:D ! ! ! dial-peer voice 10 voip destination-pattern .T session protocol sipv2 session target ipv4:IP_OF_ASTERISK:5060 session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 42 pots destination-pattern .T direct-inward-dial port 0:D ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:IP_OF_ASTERISK ! Actually, a Tcpdump on my Asterisk server don't see any trafic between asterisk and cisco and when i call a phone number that arrives on the E1, it's busy anyone have a idea ? bye jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] hi
plz do not send for me e-mail thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Calls hang up after 20 seconds
Gianni Fioretta wrote: Hello. I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason. The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and one of them answer, the call ends itself after 20 seconds from the answer. I've tried many configuration in sip.conf, but no one solved the problem. Log from /var/log/asterisk/messages: [Oct 8 15:49:05] WARNING[10659] chan_sip.c: Hanging up call e3f0204b-b35811de-bba58889-6d53c...@83.211.2.220 - no reply to our critical packet. and from CLI: [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 59e62874-b35911de-9a598915-c5a6b...@195.62.226.16 for seqno 101 (Critical Response) [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1972 retrans_pkt: Hanging up call 59e62874-b35911de-9a598915-c5a6b...@195.62.226.16 - no reply to our critical packet. == Spawn extension (incoming, 03411885583, 4) exited non-zero on 'SIP/03411885583-081e0d78' (peer 101 was not connected at this time, but Asterisk also hags up with all the peers connected) Any idea? Thanks in advance. This is a very weird Asteriskism that we see from time to time. Some SIP servers don't route ACK packets properly (or there will be an ACK loop). The nature of ACK packets is tenuous at best in the SIP world. Many clients don't even send them. Asterisk relies heavily on ACK packets to determine if a call is currently connected. If it doesn't receive one, it hangs up the call, even if the rest of the packets have been routed properly and the call is working fine. There's no configuration to turn this off, but there is a way to remove the check in the code. I can't recall the appropriate line to comment out, though. Perhaps someone else knows? In an ideal world, when Asterisk sent an ACK, whatever server/client it was connected to would respond accordingly. It is, however, not an ideal world, so this doesn't always happen. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] hi
Hello. To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users On 10/15/09, as asd sa11...@yahoo.com wrote: plz do not send for me e-mail thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Where to find IMAP storage doc ?
Hi, Where can I find doc related to IMAP storage. Usually, config options can be found either in voicemail.conf or voip-info.org but almost none relates to IMAP configuration. At the moment, I'm looking for data related to imapflags possible values. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Queues with unavailable members
That shouldn't be too hard to accomplish. If you've got the addons (and mysql) installed you could store them in a MySQL table (timestamp, device) and have a cron job set to run at X frequency that un-pauses the queue members via AMI. Don't want to go to MySQL? Use system() to 'touch' files named after devices. Then have your cron script go through the files by creation date. Either way gets you there. -Elliot -Original Message- From: Benny Amorsen [mailto:benny+use...@amorsen.dk] Sent: Thursday, October 15, 2009 5:06 AM To: Elliot Otchet Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Queues with unavailable members Elliot Otchet elliot.otc...@callingcircles.com writes: Have you tried autopause=yes in your queue configuration? You can then unpause the member by either the dialplan (e.g. having the cell phone user log back in) or using an AMI based program to change the paused state. You can read more about the latter here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html That looks very interesting, thank you! First of all though I need to avoid having them autopause just because they don't answer their phone. It should only happen if the call to their phone fails completely. I guess that could be done by not doing autopause but instead pausing manually in the context that the Local call passes through. That would also solve my second problem, which is that I need to pause it in all queues, not just one queue. The last challenge is to somehow unpause them after a while. In traditional programming that would be something like keeping a list of timeout,queuemember ordered by timeout, and then when every call comes in unpause and remove the ones where timeout expired... I'm not sure that I can make an ordered list in the dialplan though. I may have to resort to AGI, but I still need somewhere to actually store the list. Tricky. /Benny This message is intended only for the use of the individual (s) or entity to which it is addressed and may contain information that is privileged, confidential, and/or proprietary to Calling Circles LLC and its affiliates. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, forwarding or copying of this communication is prohibited without the express permission of the sender. If you have received this communication in error, please notify the sender immediately and delete the original message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Queues with unavailable members
Elliot Otchet elliot.otc...@callingcircles.com writes: That shouldn't be too hard to accomplish. If you've got the addons (and mysql) installed you could store them in a MySQL table (timestamp, device) and have a cron job set to run at X frequency that un-pauses the queue members via AMI. Don't want to go to MySQL? Use system() to 'touch' files named after devices. Then have your cron script go through the files by creation date. Either way gets you there. This seems like a very heavyweight solution. Having a cron job running every minute isn't particularly attractive, and making a daemon do the job isn't my cup of tea either. Perhaps the problem could be restated in a different way: After a queue member rejects a call (instead of just not answering), the queue should wait X amount of time before sending the next call. Queues.conf has a million settings, but I can't find one which does this. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk with a Cisco AS5300 gateway
I don't have any experience with E1, but here are some comments from the T1 perspective (on a 2800 series Cisco). Here is also a link to my collection of Cisco voice debugging commands: http://thurmantech.com/node/5 On Thu, Oct 15, 2009 at 3:27 AM, Phibee Network Operation Center n...@phibee.net wrote: ! Something like this to define what clock to use (internal usually causes a lot of Slip/Error seconds) ! but don't quote me on the following line, I haven't used E1 or the 5300 ! network-clock-select 1 E1 0 isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec preference 2 g729r8 codec preference 3 g723r63 codec preference 4 g711ulaw ! ! You don't seem to use either voice class, do you need both? voice class codec 500 codec preference 1 g729r8 codec preference 2 g723r63 ! controller E1 0 framing NO-CRC4 ! linecode ? ! cablelength ? pri-group timeslots 1-31 description E1 Beta-Test ! interface Serial0:15 no ip address encapsulation ppp isdn switch-type primary-net5 isdn incoming-voice voice no cdp enable voice-port 0:D ! ! ! dial-peer voice 10 voip destination-pattern . redirect ip2ip session protocol sipv2 session target ipv4:IP_OF_ASTERISK:5060 session transport udp dtmf-relay rtp-nte ! codec g711alaw ! If you define the codec class, might as well use it voice-class codec 400 dtmf-relay rtp-nte no vad ! dial-peer voice 42 pots ! Don't make both patterns the same, maybe add a trunk prefix here ! destination-pattern .T destination-pattern 8T incoming called-number .T direct-inward-dial port 0:D ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 ! Don't need this, since you specified it on the dial-peer ! sip-server ipv4:IP_OF_ASTERISK ! Good luck! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Where to find IMAP storage doc ?
Where can I find doc related to IMAP storage. Usually, config options can be found either in voicemail.conf or voip-info.org but almost none relates to IMAP configuration. At the moment, I'm looking for data related to imapflags possible values. More or less everything I know I found on this old email http://www.mail-archive.com/courier-us...@lists.sourceforge.net/msg27564.html It worked fine at the time although, I have to admit, I don't know if I've even tried to use it in the last 6 months -- Matthew Harrell Bit Twiddlers, Inc. mharr...@bittwiddlers.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Does ADA 1.1 or ADA Pro exists ?
Hi, Here and there, I can mentions to Asterisk Desktop Assistant versions 1.1 or Pro but I can't find any place to download or buy it. Any help ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] ChanSpy on asterisk 1.6
Thanks very much, it worked as I needed :)
On Wed, 14 Oct 2009 17:14:53 +0530, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
hey In 1.6 version actually not wrote any code for option 'o'
you need to add following line into file
Index: apps/app_chanspy.c
===
--- apps/app_chanspy.c(revision 215998)
+++ apps/app_chanspy.c(working copy)
@@ -427,7 +427,12 @@
return -1;
}
- f = ast_audiohook_read_frame(csth-spy_audiohook, samples,
AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR);
+ if (ast_test_flag(chan, OPTION_READONLY)) {
+ /* Option 'o' was set, so don't mix channel audio */
+ f = ast_audiohook_read_frame(csth-spy_audiohook, samples,
AST_AUDIOHOOK_DIRECTION_READ, AST_FORMAT_SLINEAR);
+ } else {
+ f = ast_audiohook_read_frame(csth-spy_audiohook, samples,
AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR);
+ }
ast_audiohook_unlock(csth-spy_audiohook);
regards
Dhaval
2009/10/14 Jorge Gutiérrez jgutier...@palosanto.com
I have read about that on asterisk 1.6, there will be a parameter o
(Only
listen to audio coming from this channel), I have tried, but I still get
inbound and outbound audio from the spied channel.
Has anyone used this feature? Is it working? Is there any work-around?
I will like to only spy the outbound audio from a channel, I dont want
to
hear the incomming audio of that channel.
I have used the following context:
[Conf]
exten = s,1,Answer
exten = s,2,Background(custom/menu_test)
exten = s,3,ChanSpy(,qoX)
exten = 1,1,Goto(Conf,s,2)
exten = 2,1,AGI(conf.php,${CALLERID(num)},${SPY_CHANNEL})
exten = 2,n,Goto(s,3)
exten = s,n,Goto(test2,s,1)
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--
Atentamente,
Jorge Gutiérrez
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Re: [asterisk-users] Asterisk with a Cisco AS5300 gateway
On Thu, Oct 15, 2009 at 6:27 AM, Phibee Network Operation Center n...@phibee.net wrote: dial-peer voice 10 voip destination-pattern .T session protocol sipv2 session target ipv4:IP_OF_ASTERISK:5060 session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 42 pots destination-pattern .T direct-inward-dial port 0:D What does destination-pattern .T mean? I'm not familiar with what .T would match. I would suggest using a more specific pattern that you expect to be coming down the line. Actually, a Tcpdump on my Asterisk server don't see any trafic between asterisk and cisco and when i call a phone number that arrives on the E1, it's busy Doesn't see any traffic when? When the asterisk tries to call the Cisco? That would suggest you have a sip.conf misconfiguration on asterisk. No traffic when the Cisco tries to call the asterisk? That could be for a number of reasons. I would suggest your destination-pattern could be bad, since I don't know what that syntax means. If an E1 works like a PRI T1, when you dial in, a DNIS is getting pushed to the Cisco, and that's what you should match your destination-pattern on. Regardless, SOMETHING is getting pushed down the wire when an E1 call comes in, and you're getting a busy because Cisco has no matching dial-peers. Finally, it's rather embarrassing that you're asking this from a 'network operations center' email address. How about using a personal email address with your real name? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] PSTN to SIP line ratio
Hi, I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. Regards Smir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] sporadic one-way audio
We have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice, but they can't hear us. If we immediately call them back everything is fine. The problem affects all offices and also happens when making sip to sip calls from one snom 300 to another. In addition we periodically have calls that drop off in the middle of a conversation like the connection was lost. I haven't been able to replicate any of these problems and the people that are having them can't seem to keep track of when they occur so I can go back and look in the logs. I suspect that both problems may be related though. Possibly a registration issue? Any ideas are welcome. Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] best way to make 5-10 simultaneous calls to the same did at a set time of day
I need for asterisk to call me at a predetermined number once a day at a predetermined time and once connected to me make 5-10 simultanious calls to a DID filling all available channels. What is the best way to do this? Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] PSTN to SIP line ratio
Shahnawaz Mir wrote: I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. I don't think there is an answer, it really depends on what your 200 users are going to do. It is not the same if half of your 200 users are sales reps, or if you are setting up 200 extensions for a school where the calls are going to be mostly internal... -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] PSTN to SIP line ratio
On Thu, 15 Oct 2009, Shahnawaz Mir wrote: I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. 42[:1] (The fact that you ask such a generic question implies you have a high probability of failure. You should hire somebody with a bit more clue and learn from them.) -- 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 -- AstriCon 2009 - October 13 - 15 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] best way to make 5-10 simultaneous calls to the same did at a set time of day
On Thu, 15 Oct 2009, Eric Fort wrote: I need for asterisk to call me at a predetermined number once a day at a predetermined time and once connected to me make 5-10 simultanious calls to a DID filling all available channels. What is the best way to do this? Eric What's best for you may not be best for me. How will you know how many calls to make? Probably the easiest (requiring the least knowledge and labor) would be to use cron to schedule a script that would create a call file in /tmp/ and then move it to Asterisk's outgoing directory. Your call file would call you and then jump into your dialplan where you would use system() to create the remaining call files and then dump you into a meetme conference. The remaining call files would call the DID and then jump into your dialplan where you would dump the calls into your meetme. If you have the skills, AMI (scheduled by cron) would probably be best. -- 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 -- AstriCon 2009 - October 13 - 15 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] PSTN to SIP line ratio
- Steve Edwards asterisk@sedwards.com wrote: On Thu, 15 Oct 2009, Shahnawaz Mir wrote: I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. 42[:1] (The fact that you ask such a generic question implies you have a high probability of failure. You should hire somebody with a bit more clue and learn from them.) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Ignoring unhelpful snobbish remarks from the peanut gallery... Your ratio will depend largely on the usage by your users. In a busy contact center where your users/agents will be on calls nearly 100% of the time, your ratio will need to be closer to 1:1. However, if the installation is for a school where most of the staff (teachers) are instructing in the classroom or otherwise away from their desks, you can get by with a higher ratio like 4:1. As always, you build your system with room for expansion in the event you need additional resource availability. Also, ensure your customer/client understands the limitations of the number simultaneous calls. If you don't tell them and they find out the hard way, you'll be in a world of hurt. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] PSTN to SIP line ratio
Shahnawaz Mir wrote: Hi, I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. Regards Smir _ You need to undertand traffic. See for instance: http://www.wirelesscommunication.nl/reference/chaptr04/erlang/erlang.htm Regards Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser= morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Where to find IMAP storage doc ?
2009/10/15 Matthew Harrell lists-sender-6a8...@bittwiddlers.com Where can I find doc related to IMAP storage. Usually, config options can be found either in voicemail.conf or voip-info.org but almost none relates to IMAP configuration. At the moment, I'm looking for data related to imapflags possible values. More or less everything I know I found on this old email http://www.mail-archive.com/courier-us...@lists.sourceforge.net/msg27564.html It worked fine at the time although, I have to admit, I don't know if I've even tried to use it in the last 6 months It seems very informative ! Thanks for the tip. PS: Which c-client library should be used ? This doc says to compile from source but a 2007 version exist in Lenny. -- Matthew Harrell Bit Twiddlers, Inc. mharr...@bittwiddlers.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
Just a thought... If the SNOM has multiple lines, tying one to 612 and the other to 610 should make the MWI active for both lines. Asterisk AFAIK only actives the first entry in the list, so you would need two entries for tkeeley with mailbox=612 in the first instance and mailbox=610 in the second. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] PSTN to SIP line ratio
Thanks Tim, Your response is really helpful. Its not going to be very busy. I was expecting 10:1 but I will start some where between 4-10. Thank you very much. Regards Shahnawaz Mir On 15-Oct-09, at 11:11 AM, Tim Nelson wrote: - Steve Edwards asterisk@sedwards.com wrote: On Thu, 15 Oct 2009, Shahnawaz Mir wrote: I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. 42[:1] (The fact that you ask such a generic question implies you have a high probability of failure. You should hire somebody with a bit more clue and learn from them.) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Ignoring unhelpful snobbish remarks from the peanut gallery... Your ratio will depend largely on the usage by your users. In a busy contact center where your users/agents will be on calls nearly 100% of the time, your ratio will need to be closer to 1:1. However, if the installation is for a school where most of the staff (teachers) are instructing in the classroom or otherwise away from their desks, you can get by with a higher ratio like 4:1. As always, you build your system with room for expansion in the event you need additional resource availability. Also, ensure your customer/client understands the limitations of the number simultaneous calls. If you don't tell them and they find out the hard way, you'll be in a world of hurt. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] ChanSpy
No, Extenspy was introduced in 1.4 as far as I know.
Chanspy is simple :) Helpful as I am, I'm gonna paste here the output of show
application chanspy
callcenter*CLI show application ChanSpy
callcenter*CLI
-= Info about application 'ChanSpy' =-
[Synopsis]
Listen to a channel, and optionally whisper into it
[Description]
ChanSpy([chanprefix][|options]): This application is used to listen to the
audio from an Asterisk channel. This includes the audio coming in and
out of the channel being spied on. If the 'chanprefix' parameter is specified,
only channels beginning with this string will be spied upon.
While spying, the following actions may be performed:
- Dialing # cycles the volume level.
- Dialing * will stop spying and look for another channel to spy on.
- Dialing a series of digits followed by # builds a channel name to append
to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing
the digits '1234#' while spying will begin spying on the channel
'Agent/1234'.
Options:
b - Only spy on channels involved in a bridged call.
g(grp)- Match only channels where their ${SPYGROUP} variable is set
to
contain 'grp' in an optional : delimited list.
q - Don't play a beep when beginning to spy on a channel, or
speak the
selected channel name.
r[(basename)] - Record the session to the monitor spool directory. An
optional base for the filename may be specified. The
default is 'chanspy'.
v([value])- Adjust the initial volume in the range from -4 to 4. A
negative value refers to a quieter setting.
w - Enable 'whisper' mode, so the spying channel can talk to
the spied-on channel.
W - Enable 'private whisper' mode, so the spying channel can
talk to the spied-on channel but cannot listen to that
channel.
Hope you get it to work. Also http://voip-info.org is a great source of
information about asterisk. Good luck
Regards
Rennes Neps
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of Torintino T
Sent: Wed 10/14/2009 22:48
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ChanSpy
Thanks for your reply.
Is ExtenSpy available in Asterisk 1.2?
If yes, please how can i use it?
and how can i cycle through the available channels by ChanSpy?
Thanks.
Torintino
Date: Wed, 14 Oct 2009 18:15:29 +0300
From: rennes.n...@norby.ee
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] ChanSpy
You must use extenspy if you want to spy on specific extension. Otherwise you
can only cycle through available channels.
Regards
Rennes
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of Torintino T
Sent: Wed 10/14/2009 17:46
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ChanSpy
I am unsing Asterisk 1.2.28
I want please to use ChanSpy urgently
my /etc/asterisk/extensions_additional.conf is as follow:
[chanspy]
include = chanspy-custom
exten = 102**,1,Chanspy(102)
exten = 102**,n,Hangup
exten = 103**,1,Chanspy(103)
exten = 103**,n,Hangup
exten = 400**,1,Chanspy(400)
exten = 400**,n,Hangup
exten = 501**,1,Chanspy(501)
exten = 501**,n,Hangup
exten = 601**,1,Chanspy(601)
exten = 601**,n,Hangup
exten = 606**,1,Chanspy(606)
exten = 606**,n,Hangup
; end of [chanspy]
I created a Context to put my extension into it to be able to use ChanSpy.
While there is a call with an extension 102 and my extension is 606
i call 102** to spy but i couldn't hear anything, all i hear is beep
-- Executing ChanSpy(SIP/606-09430fd0, 102) in new stack
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
Thanks
Torintino
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Re: [asterisk-users] Where to find IMAP storage doc ?
On Thu, 2009-10-15 at 10:52 -0400, Matthew Harrell wrote: Where can I find doc related to IMAP storage. Usually, config options can be found either in voicemail.conf or voip-info.org but almost none relates to IMAP configuration. At the moment, I'm looking for data related to imapflags possible values. More or less everything I know I found on this old email http://www.mail-archive.com/courier-us...@lists.sourceforge.net/msg27564.html It worked fine at the time although, I have to admit, I don't know if I've even tried to use it in the last 6 months We're also working fine with it but I also do not know what the available imapflags are and what they mean. I have seen notls and novalidatecert. Out of curiosity, I spent the last 20 minutes googling for information on c-client imapflags and didn't find any definitions or even a simple list, either. There is a list of flags in the c-client man page but they seem to be a different set of flags. Let me know what you find as I would like to know what functionality and options they give us. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
Ah, interesting. I wasn't aware that it only used the first value. What's the purpose of the secondary values then? If I understand you correctly, you are saying I should have one entry for tkeeley with two entries for mailbox=? Thanks - John On Thu, 2009-10-15 at 12:54 -0500, Danny Nicholas wrote: Just a thought... If the SNOM has multiple lines, tying one to 612 and the other to 610 should make the MWI active for both lines. Asterisk AFAIK only actives the first entry in the list, so you would need two entries for tkeeley with mailbox=612 in the first instance and mailbox=610 in the second. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
The secondary value is used, just not by the MWI functionality. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Ah, interesting. I wasn't aware that it only used the first value. What's the purpose of the secondary values then? If I understand you correctly, you are saying I should have one entry for tkeeley with two entries for mailbox=? Thanks - John On Thu, 2009-10-15 at 12:54 -0500, Danny Nicholas wrote: Just a thought... If the SNOM has multiple lines, tying one to 612 and the other to 610 should make the MWI active for both lines. Asterisk AFAIK only actives the first entry in the list, so you would need two entries for tkeeley with mailbox=612 in the first instance and mailbox=610 in the second. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
No, I'm saying you need two tkeeley entries with one mailbox each. The multiple entry is fine for other mailbox functionality, just not MWI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Ah, interesting. I wasn't aware that it only used the first value. What's the purpose of the secondary values then? If I understand you correctly, you are saying I should have one entry for tkeeley with two entries for mailbox=? Thanks - John On Thu, 2009-10-15 at 12:54 -0500, Danny Nicholas wrote: Just a thought... If the SNOM has multiple lines, tying one to 612 and the other to 610 should make the MWI active for both lines. Asterisk AFAIK only actives the first entry in the list, so you would need two entries for tkeeley with mailbox=612 in the first instance and mailbox=610 in the second. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Queues with unavailable members
At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
Perhaps the problem could be restated in a different way: After a
queue member rejects a call (instead of just not answering), the
queue should wait X amount of time before sending the next call.
Queues.conf has a million settings, but I can't find one which does
this.
To pause an agent, store the unpause time per agent in the AstDB.
Then when you're deciding whether to give out a call (in the Local
channel), look up ${DB(AgentPaused/agentid)} and compare it to the
current time. If there is no record or the time has passed, put the
call through; otherwise, skip that agent.
Sorry, no example code yet... I just wanted to get the idea out there.
--
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The Lodging Company
http://www.skihills.com/
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[asterisk-users] Testing the Timing Device
Hello, Does anyone know how to test the timing device? I've tried the following but with no luck. Zaptel is installed. I'm trying to use ztdummy as a timer. [r...@templateasteriskserver ~]# dahdi_test Unable to open dahdi interface: No such file or directory [r...@templateasteriskserver ~]# zttool Unable to open /dev/zap/ctl: No such file or directory Thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] no outbound calls
You have to check and verify the SIP trunk details, as ext to ext works once the pbx is up, but to call out, it should go through your provider.so just recheck your provider's details. Regards Sandesh On Wed, Oct 14, 2009 at 8:24 AM, Ott Rose sixfourimp...@hotmail.com wrote: here is the debug from the CLI. I think I know where the problem is I just can figure out how to fix it. The IP in the From and To i think is where the problem is. When I make an outbound call. i get the message the call cannot be completed as dialed. if i call another ext it works. I posted the debug for both calls. ==outbound call=== --- Transmitting (NAT) to 10.0.0.46:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.46:5060 ;branch=z9hG4bKfd2143ede5319cf9b.273e4b904b0cc3101;received=10.0.0.46 From: ext sip:1...@10.0.0.8 sip%3a...@10.0.0.8;tag=9d9e3944ba To: 93214545 sip:93214...@10.0.0.8 sip%3a93214...@10.0.0.8 ;tag=as290bd498 Call-ID: 401d30b0a1893e80 CSeq: 13401 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:99676...@10.0.0.8 sip%3a99676...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 14398 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv = ext to ext=== SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.46:5060 ;branch=z9hG4bK280378de3608b5bf1.2576ced30c6198b74;received=10.0.0.46 From: ext sip:1...@10.0.0.8 sip%3a...@10.0.0.8;tag=d729237fcc To: 111 sip:1...@10.0.0.8 sip%3a...@10.0.0.8;tag=as553ab5e9 Call-ID: c7cc32657c620790 CSeq: 8007 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:1...@10.0.0.8 sip%3a...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 10414 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. http://clk.atdmt.com/GBL/go/177141664/direct/01/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Queues with unavailable members
At 11:32 AM on 15 Oct 2009, C. Chad Wallace wrote:
At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
Perhaps the problem could be restated in a different way: After a
queue member rejects a call (instead of just not answering), the
queue should wait X amount of time before sending the next call.
Queues.conf has a million settings, but I can't find one which does
this.
To pause an agent, store the unpause time per agent in the AstDB.
Then when you're deciding whether to give out a call (in the Local
channel), look up ${DB(AgentPaused/agentid)} and compare it to the
current time. If there is no record or the time has passed, put the
call through; otherwise, skip that agent.
Sorry, no example code yet... I just wanted to get the idea out
there.
OK, I decided to write it up in AEL. It's incomplete and untested, but
it probably gets the idea across a little better.
context agentcalls {
_2XX = {
Set(AGENT=${EXTEN}); // Assuming agent ID is extension.
if (${EPOCH}${DB(AgentPaused/${AGENT})}) {
// Let the call through to the cell phone
Dial(...);
if (cell call was rejected) {
// Flag agent as paused for the next 30 seconds.
Set(DB(AgentPaused/${AGENT})=$[${EPOCH}+30]);
};
}
else {
// Agent still paused.
};
};
};
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
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Re: [asterisk-users] Testing the Timing Device
You need to do /etc/init.d/dahdi start or /etc/init.d/zaptel start to load the devices or dummy devices _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, October 15, 2009 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Testing the Timing Device Hello, Does anyone know how to test the timing device? I've tried the following but with no luck. Zaptel is installed. I'm trying to use ztdummy as a timer. [r...@templateasteriskserver ~]# dahdi_test Unable to open dahdi interface: No such file or directory [r...@templateasteriskserver ~]# zttool Unable to open /dev/zap/ctl: No such file or directory Thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best way to detect fax in Asterisk 1.6??
Hello, I´ve found information about NVFax, app_fax, NVBackgroundDetect, rxfax, etc But which is the best way for *detecting fax in Asterisk 1.6*??? I will use it in an automatic dialer. Thank you very much, Pablo Bernasconi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] DS3 capacity calls using asterisk
Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] OT - Can't upgrade Cisco 7942 to SIP
Hi, I've downloaded for a demo, a P0S3-08-12.zip file which is suppose to work with 7960. Is it supposed to be the same file that the one needed to 7942 model ? At the moment, my 7942 is blocked when trying to download a P0S3-8-12-00.loads file. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] DS3 capacity calls using asterisk
das sandesh sandesh...@gmail.com wrote: Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh Putting aside the obvious question of WHY ON EARTH WOULD YOU WANT ALMOST 700 CALLS HANDLED ON **ONE** ASTERISK BOX... ... Sangoma makes a DS3 card but it is not channelized for voice usage. We had a conversation with them at one point for a particular project and if I recall they do have a solution for running channelized voice over DS3. Give them a call or email, they are incredibly responsive and 'know their stuff'. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Testing the Timing Device
Hi Danny, I've tried that but I get the following errors:- [r...@templateasteriskserver ~]# /etc/init.d/dahdi start Loading DAHDI hardware modules: FATAL: Module dahdi not found. wct4xxp: FATAL: Module wct4xxp not found. [FAILED] wcte12xp: FATAL: Module wcte12xp not found. [FAILED] wct1xxp: FATAL: Module wct1xxp not found. [FAILED] wcte11xp: FATAL: Module wcte11xp not found. [FAILED] wctdm24xxp: FATAL: Module wctdm24xxp not found. [FAILED] wcfxo: FATAL: Module wcfxo not found. [FAILED] wctdm: FATAL: Module wctdm not found. [FAILED] wcb4xxp: FATAL: Module wcb4xxp not found. [FAILED] wctc4xxp: FATAL: Module wctc4xxp not found. [FAILED] xpp_usb: FATAL: Module xpp_usb not found. [FAILED] [r...@templateasteriskserver ~]# /etc/init.d/zaptel start Loading zaptel framework: FATAL: Module zaptel not found. [FAILED] Waiting for zap to come online... [r...@templateasteriskserver ~]# Any ideas? Thanks Dan Journo Thank you for contacting Kesher Communications Ltd. IT Maintenance Clients can now receive a faster response by using our Live Chat and Support Service: Click Here http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7 F508C6B61A91700343lang=ensurpre=PreSurvey This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. Warning: Although the Company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. All prices exclude VAT unless otherwise stated. No responsibility is taken for any recommendations made by a sender or by Kesher Communications Ltd. Recipient(s) takes responsibility for any actions taken as a result of advice and recommendations given by Kesher Communications Ltd. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 15 October 2009 19:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Testing the Timing Device You need to do /etc/init.d/dahdi start or /etc/init.d/zaptel start to load the devices or dummy devices From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, October 15, 2009 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Testing the Timing Device Hello, Does anyone know how to test the timing device? I've tried the following but with no luck. Zaptel is installed. I'm trying to use ztdummy as a timer. [r...@templateasteriskserver ~]# dahdi_test Unable to open dahdi interface: No such file or directory [r...@templateasteriskserver ~]# zttool Unable to open /dev/zap/ctl: No such file or directory Thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Testing the Timing Device
What does /etc/dahdi/modules look like? I suspect that it has each of the wc* entries in it. If so, remove those lines and put in dummy (just once). _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, October 15, 2009 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Testing the Timing Device Hi Danny, I've tried that but I get the following errors:- [r...@templateasteriskserver ~]# /etc/init.d/dahdi start Loading DAHDI hardware modules: FATAL: Module dahdi not found. wct4xxp: FATAL: Module wct4xxp not found. [FAILED] wcte12xp: FATAL: Module wcte12xp not found. [FAILED] wct1xxp: FATAL: Module wct1xxp not found. [FAILED] wcte11xp: FATAL: Module wcte11xp not found. [FAILED] wctdm24xxp: FATAL: Module wctdm24xxp not found. [FAILED] wcfxo: FATAL: Module wcfxo not found. [FAILED] wctdm: FATAL: Module wctdm not found. [FAILED] wcb4xxp: FATAL: Module wcb4xxp not found. [FAILED] wctc4xxp: FATAL: Module wctc4xxp not found. [FAILED] xpp_usb: FATAL: Module xpp_usb not found. [FAILED] [r...@templateasteriskserver ~]# /etc/init.d/zaptel start Loading zaptel framework: FATAL: Module zaptel not found. [FAILED] Waiting for zap to come online... [r...@templateasteriskserver ~]# Any ideas? Thanks Dan Journo _ Thank you for contacting Kesher Communications Ltd. IT Maintenance Clients can now receive a faster response by using our Live Chat and Support Service: Click http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508 C6B61A91700343lang=ensurpre=PreSurvey Here This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. Warning: Although the Company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. All prices exclude VAT unless otherwise stated. No responsibility is taken for any recommendations made by a sender or by Kesher Communications Ltd. Recipient(s) takes responsibility for any actions taken as a result of advice and recommendations given by Kesher Communications Ltd. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 15 October 2009 19:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Testing the Timing Device You need to do /etc/init.d/dahdi start or /etc/init.d/zaptel start to load the devices or dummy devices _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, October 15, 2009 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Testing the Timing Device Hello, Does anyone know how to test the timing device? I've tried the following but with no luck. Zaptel is installed. I'm trying to use ztdummy as a timer. [r...@templateasteriskserver ~]# dahdi_test Unable to open dahdi interface: No such file or directory [r...@templateasteriskserver ~]# zttool Unable to open /dev/zap/ctl: No such file or directory Thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] A little OT but need an opinion on Aastra 57i CT
Hello All, I have a need for a wireless solution and have been looking at the Aastra 57i CT phone that have the wireless handset with them. Aastra says they will cover up to 300,000 square feet. I am finding this hard to accept. I was also wondering about the secure WDCT cordless technology Could this be a form of DECT? Any one using these that can shed some lite? Thanks. JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Testing the Timing Device
Ok, its a little better now. But I still get a fatal message:- [r...@templateasteriskserver dahdi]# /etc/init.d/dahdi start Loading DAHDI hardware modules: FATAL: Module dahdi not found. dummy: [ OK ] Any ideas? Thanks Dan Thank you for contacting Kesher Communications Ltd. IT Maintenance Clients can now receive a faster response by using our Live Chat and Support Service: Click Here http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7 F508C6B61A91700343lang=ensurpre=PreSurvey This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. Warning: Although the Company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. All prices exclude VAT unless otherwise stated. No responsibility is taken for any recommendations made by a sender or by Kesher Communications Ltd. Recipient(s) takes responsibility for any actions taken as a result of advice and recommendations given by Kesher Communications Ltd. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 15 October 2009 20:40 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Testing the Timing Device What does /etc/dahdi/modules look like? I suspect that it has each of the wc* entries in it. If so, remove those lines and put in dummy (just once). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, October 15, 2009 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Testing the Timing Device Hi Danny, I've tried that but I get the following errors:- [r...@templateasteriskserver ~]# /etc/init.d/dahdi start Loading DAHDI hardware modules: FATAL: Module dahdi not found. wct4xxp: FATAL: Module wct4xxp not found. [FAILED] wcte12xp: FATAL: Module wcte12xp not found. [FAILED] wct1xxp: FATAL: Module wct1xxp not found. [FAILED] wcte11xp: FATAL: Module wcte11xp not found. [FAILED] wctdm24xxp: FATAL: Module wctdm24xxp not found. [FAILED] wcfxo: FATAL: Module wcfxo not found. [FAILED] wctdm: FATAL: Module wctdm not found. [FAILED] wcb4xxp: FATAL: Module wcb4xxp not found. [FAILED] wctc4xxp: FATAL: Module wctc4xxp not found. [FAILED] xpp_usb: FATAL: Module xpp_usb not found. [FAILED] [r...@templateasteriskserver ~]# /etc/init.d/zaptel start Loading zaptel framework: FATAL: Module zaptel not found. [FAILED] Waiting for zap to come online... [r...@templateasteriskserver ~]# Any ideas? Thanks Dan Journo Thank you for contacting Kesher Communications Ltd. IT Maintenance Clients can now receive a faster response by using our Live Chat and Support Service: Click Here http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7 F508C6B61A91700343lang=ensurpre=PreSurvey This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. Warning: Although the Company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. All prices exclude VAT unless otherwise stated. No responsibility is taken for any recommendations made by a sender or by Kesher Communications Ltd. Recipient(s) takes responsibility for any actions taken as a result of advice and recommendations given by Kesher Communications Ltd. From: asterisk-users-boun...@lists.digium.com
Re: [asterisk-users] PSTN to SIP line ratio
On Thu, 15 Oct 2009, Shahnawaz Mir wrote: I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. - Steve Edwards asterisk@sedwards.com wrote: 42[:1] (The fact that you ask such a generic question implies you have a high probability of failure. You should hire somebody with a bit more clue and learn from them.) On Thu, 15 Oct 2009, Tim Nelson wrote: Ignoring unhelpful snobbish remarks from the peanut gallery... Really? Seriously? Maybe facetious and direct, but not intended to be snobbish or unhelpful. Your ratio will depend largely on the usage by your users. In a busy contact center where your users/agents will be on calls nearly 100% of the time, your ratio will need to be closer to 1:1. However, if the installation is for a school where most of the staff (teachers) are instructing in the classroom or otherwise away from their desks, you can get by with a higher ratio like 4:1. Assuming the OP meant how many PSTN lines (incoming and outgoing) for every x SIP users (and that you meant 1:4), without knowing the application, 42:1 (or 1:42) is just as relevant as 1:1 or 1:4. (As a parent, if 25% of the teachers are yakking to somebody off-campus, I'd raise holy hell.) As always, you build your system with room for expansion in the event you need additional resource availability. Also, ensure your customer/client understands the limitations of the number simultaneous calls. If you don't tell them and they find out the hard way, you'll be in a world of hurt. Without the guidance of someone who has been down the road before, he is likely to hit every pothole. I still think hiring some talent to either do the job or mentor the OP is the most helpful advice I could give. -- 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 -- AstriCon 2009 - October 13 - 15 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] 2 IPs for an Asterisk server.
Hello. I've been setting up an Asterisk server, and I am now supposed to move it to a different network than the one it was set on. I'd like to give the server 2 IP address: -1- The first IP address is the IP it will have on the LAN, meaning that softphones will register to the Asterisk server using this 1st IP. -2- The second IP is the one that it will use to connect to the remote VoIP provider, which is using another network range than the LAN where I have my softphones. The default gateway would be the one of this second network address range. No NAT involved anywhere in this setup. Is it possible to do such a thing with Asterisk? Does it need really special tweaking of Asterisk conf files? -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Testing the Timing Device
Try this thread http://forums.digium.com/viewtopic.php?p=132042sid=297f2470a0a3d87e91efc1a5 9defcab9 _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, October 15, 2009 3:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Testing the Timing Device Ok, its a little better now. But I still get a fatal message:- [r...@templateasteriskserver dahdi]# /etc/init.d/dahdi start Loading DAHDI hardware modules: FATAL: Module dahdi not found. dummy: [ OK ] Any ideas? Thanks Dan _ Thank you for contacting Kesher Communications Ltd. IT Maintenance Clients can now receive a faster response by using our Live Chat and Support Service: Click http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508 C6B61A91700343lang=ensurpre=PreSurvey Here This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. Warning: Although the Company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. All prices exclude VAT unless otherwise stated. No responsibility is taken for any recommendations made by a sender or by Kesher Communications Ltd. Recipient(s) takes responsibility for any actions taken as a result of advice and recommendations given by Kesher Communications Ltd. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] A little OT but need an opinion on Aastra 57i CT
At 12:50 PM 10/15/2009, you wrote: I have a need for a wireless solution and have been looking at the Aastra 57i CT phone that have the wireless handset with them. Aastra says they will cover up to 300,000 square feet. I am finding this hard to accept. I was also wondering about the secure WDCT cordless technology Could this be a form of DECT? We have 3 of the 480i-CT in our 3 bedroom townhouse. They work fine, but will not seem to make it more than 50 or 100' from the base station. Great phone and the cordless option is just wonderful, but limited. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] 2 IPs for an Asterisk server.
Yes it is possible, the only thing that you need to do is to configure correctly your network routes, if your ip devices are on the same net of your elastix you wont need to do any route configuration. Just leave the default gateway for your wan provider, it should work without any trouble On Thu, 15 Oct 2009 21:58:47 +0200, Guillaume Yziquel guillaume.yziq...@citycable.ch wrote: Hello. I've been setting up an Asterisk server, and I am now supposed to move it to a different network than the one it was set on. I'd like to give the server 2 IP address: -1- The first IP address is the IP it will have on the LAN, meaning that softphones will register to the Asterisk server using this 1st IP. -2- The second IP is the one that it will use to connect to the remote VoIP provider, which is using another network range than the LAN where I have my softphones. The default gateway would be the one of this second network address range. No NAT involved anywhere in this setup. Is it possible to do such a thing with Asterisk? Does it need really special tweaking of Asterisk conf files? -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Atentamente, Jorge Gutiérrez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] A little OT but need an opinion on Aastra 57i CT
The 57i and 480i are good wireless phones but after 100ft you are out of range (assuming business interiors). Of you still have to deal with buggy firmware(and hit and miss tech support). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Millican Sent: Thursday, October 15, 2009 3:51 PM To: Asterisk Users List Subject: [asterisk-users] A little OT but need an opinion on Aastra 57i CT Hello All, I have a need for a wireless solution and have been looking at the Aastra 57i CT phone that have the wireless handset with them. Aastra says they will cover up to 300,000 square feet. I am finding this hard to accept. I was also wondering about the secure WDCT cordless technology Could this be a form of DECT? Any one using these that can shed some lite? Thanks. JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] PSTN to SIP line ratio
- Steve Edwards asterisk@sedwards.com wrote: - Steve Edwards asterisk@sedwards.com wrote: 42[:1] (The fact that you ask such a generic question implies you have a high probability of failure. You should hire somebody with a bit more clue and learn from them.) On Thu, 15 Oct 2009, Tim Nelson wrote: Ignoring unhelpful snobbish remarks from the peanut gallery... Really? Seriously? Maybe facetious and direct, but not intended to be snobbish or unhelpful. Your ratio will depend largely on the usage by your users. In a busy contact center where your users/agents will be on calls nearly 100% of the time, your ratio will need to be closer to 1:1. However, if the installation is for a school where most of the staff (teachers) are instructing in the classroom or otherwise away from their desks, you can get by with a higher ratio like 4:1. Assuming the OP meant how many PSTN lines (incoming and outgoing) for every x SIP users (and that you meant 1:4), without knowing the application, 42:1 (or 1:42) is just as relevant as 1:1 or 1:4. (As a parent, if 25% of the teachers are yakking to somebody off-campus, I'd raise holy hell.) As always, you build your system with room for expansion in the event you need additional resource availability. Also, ensure your customer/client understands the limitations of the number simultaneous calls. If you don't tell them and they find out the hard way, you'll be in a world of hurt. Without the guidance of someone who has been down the road before, he is likely to hit every pothole. I still think hiring some talent to either do the job or mentor the OP is the most helpful advice I could give. I'm certainly not saying the OP couldn't use some experienced help or consultation, I just thought the message came across as 'Go away NEWB we don't want to help'. Also, it seems rather apparent your choice of 42 is related more to http://en.wikipedia.org/wiki/42_(number)#In_The_Hitchhiker.27s_Guide_to_the_Galaxy than it is to channel ratios... While certainly amusing, it is not helpful. Anyways, it's just a mailing list, just software, just one's and zero's. Moving along... :-) --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] DS3 capacity calls using asterisk
On Thu, 15 Oct 2009, das sandesh wrote: We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? 672 eggs in 1 basket doesn't sound like a good plan to me. It's a bit out of my league (300 is my biggest installation so far), but I'd suspect dedicated hardware like something from Cisco or a used Ascend TNT to take the DS3 in 1 side and spit SIP over Ethernet out the other would be a good starting point. Send the SIP calls out to a couple of OpenSER/OpenSIPS/Kamailio servers talking to several (like 4+) Asterisk servers. This way you can take individual servers out of production without disrupting everything at once. My 300 concurrent call project was taking SIP calls from a Tekalec 7000 talking to 4 hosts. Each host ran OpenSER on port 5060 and Asterisk on port 5061. The client loved the flexibility. -- 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 -- AstriCon 2009 - October 13 - 15 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] 2 IPs for an Asterisk server.
Jorge Gutiérrez a écrit : Yes it is possible, the only thing that you need to do is to configure correctly your network routes, if your ip devices are on the same net of your elastix you wont need to do any route configuration. Just leave the default gateway for your wan provider, it should work without any trouble Thank you for this valuable information. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk with a Cisco AS5300 gateway
Hi Men, I believe that .T is anything + a Time out of (probably) 3 sec. before to dial the complete called number. Best Regards, Francois destination-pattern .T What does destination-pattern .T mean? I'm not familiar with what .T would match. I would suggest using a more specific pattern that you expect to be coming down the line. Ce message sortant est certifié sans virus connu. Analyse effectuée par AVG - www.avg.fr Version: 8.5.421 / Base de données virale: 270.14.18/2437 - Date: 10/15/09 03:57:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] DS3 capacity calls using asterisk
No usable DS3 cards for Asterisk. There is a standing consensus, as far as I've been able to tell (and I could be wrong), that this would be rather difficult - if not impossible - to do given the liberal timing tolerance of PCI buses and PC architecture once you're talking about that much synchronous payload and framing. There would be other problems, one of which is that it's not generally possible to handle that many calls concurrently with Asterisk on a single machine. Your best bet is to break a bunch of T1s out of an M13 mux, if you're determined to do this the TDM way. I'm not sure I see the point, personally; it's not because I think everyone should be using SIP origination (quite the contrary, it's still so broken in so many ways industrially), I just think that by the time it gets to Asterisk it can stand to be SIP. Get an ISDN-VoIP gateway that can take a DS3 and spit SIP out the back side and you're golden. Tim Nelson wrote: das sandesh sandesh...@gmail.com wrote: Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh Putting aside the obvious question of WHY ON EARTH WOULD YOU WANT ALMOST 700 CALLS HANDLED ON **ONE** ASTERISK BOX.. Sangoma makes a DS3 card but it is not channelized for voice usage. We had a conversation with them at one point for a particular project and if I recall they do have a solution for running channelized voice over DS3. Give them a call or email, they are incredibly responsive and 'know their stuff'. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- 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 -- AstriCon 2009 - October 13 - 15 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] DS3 capacity calls using asterisk
On Thu, Oct 15, 2009 at 4:58 PM, David Backeberg dbackeb...@gmail.com wrote: There's no one-step solution I'm aware of. Cisco sells something called an AS5300 that supposedly can terminate a DS3 and convert it all to SIP. Otherwise, you need a channel bank like the Adtran MX2800 I was close, but incorrect. Cisco sells the 5XXX series, but I think the AS5300 has a lower capacity that a full DS3. The 58xx series claims to terminate multiple DS3s. I've never played with anything nicer than a Cisco 3845, which maxes out at 24T1s, just shy of what you can get out of the Adtran MX 2800. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Where to find IMAP storage doc ?
We're also working fine with it but I also do not know what the available imapflags are and what they mean. I have seen notls and novalidatecert. Out of curiosity, I spent the last 20 minutes googling for information on c-client imapflags and didn't find any definitions or even a simple list, either. There is a list of flags in the c-client man page but they seem to be a different set of flags. Let me know what you find as I would like to know what functionality and options they give us. Thanks - John Yeah, that's about the extent of what I found with google and one of the wikis a while back. I kind of wish I could find more. The definitive answer would be to look in the source but that doesn't necessarily explain what they do -- Matthew Harrell Bit Twiddlers, Inc. mharr...@bittwiddlers.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] DS3 capacity calls using asterisk
On Thu, Oct 15, 2009 at 3:20 PM, das sandesh sandesh...@gmail.com wrote: Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? There's no one-step solution I'm aware of. Cisco sells something called an AS5300 that supposedly can terminate a DS3 and convert it all to SIP. Otherwise, you need a channel bank like the Adtran MX2800 to break the DS3 into 28 T1s, which you can then terminate in your preferred manner, either into appliances or actual PC hardware with Digium cards. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] DS3 capacity calls using asterisk
David Backeberg wrote: On Thu, Oct 15, 2009 at 4:58 PM, David Backeberg dbackeb...@gmail.com wrote: There's no one-step solution I'm aware of. Cisco sells something called an AS5300 that supposedly can terminate a DS3 and convert it all to SIP. Otherwise, you need a channel bank like the Adtran MX2800 I was close, but incorrect. Cisco sells the 5XXX series, but I think the AS5300 has a lower capacity that a full DS3. The 58xx series claims to terminate multiple DS3s. I've never played with anything nicer than a Cisco 3845, which maxes out at 24T1s, just shy of what you can get out of the Adtran MX 2800. Yes, the AS5300 chassis can only do 4 T1s. You're looking for an AS5400, or another big router chassis that can take a DS3 adaptor and VFCs (like a 7200). -- 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 -- AstriCon 2009 - October 13 - 15 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] question on SIP and call manager
Customer has 2 call manager systems and I am using asterisk to place calls through the CCM. One for the main use - CCMMAIN and another for disaster CCMSLAVE. Can asterisk be setup in such a way that calls first try to use CCMMAIN and if thats not available use CCMSLAVE. Example if I place a call file that places a call like Dial: SIP/CCMMAIN/5551212 that if CCMMAIN is not available then CCMSLAVE will automatically be used? My application placing calls in the call file doesnt have any knowledge of which context to use. CCMMAIN is the only thing my call file nows about. How do I set up such an arrangement if possible? thanks. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] question on SIP and call manager
Here are two ways to address this 1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once 2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt) Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt) CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3 rings) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, October 15, 2009 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on SIP and call manager Customer has 2 call manager systems and I am using asterisk to place calls through the CCM. One for the main use - CCMMAIN and another for disaster CCMSLAVE. Can asterisk be setup in such a way that calls first try to use CCMMAIN and if thats not available use CCMSLAVE. Example if I place a call file that places a call like Dial: SIP/CCMMAIN/5551212 that if CCMMAIN is not available then CCMSLAVE will automatically be used? My application placing calls in the call file doesnt have any knowledge of which context to use. CCMMAIN is the only thing my call file nows about. How do I set up such an arrangement if possible? thanks. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Asterisk in the Cloud
On Oct 14, 2009, at 1:04 AM, Dan Journo wrote: Thanks Eric, I'd love to be able to make it to an Astricon one day. At the moment, its a bit out of my price range. Do you happen to know whether RackspaceCloud.com offers a Kernel with a timing device enabled? Many thanks and good luck with the presentation. Dan Dan, I'm not sure what Rackspace Cloud offers kernel wise. We didn't go with them because of their higher bandwidth costs and all the other services Amazon offers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk and FreePBX Amazon EC2 instances are now available in Europe
Based on interest expressed at AstriCon, we've published Asterisk and FreePBX Amazon EC2 instances in Europe (previously they were only available in the U.S. region). More information is available at: http://voxilla.com/2009/10/15/asterisk-on-the-cloud-with-a-click-1405 http://voxilla.com/2009/10/15/freepbx-in-a-cloud-with-a-click-1436 -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] OT wanted old Sipura firmware 2.0.13
Does anybody know here I can find old Sipura firmware 2.0.13 for SPA-3000 I have Cisco 3.1.20 but it is not working as it suppose to. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Testing the Timing Device
On Thu, Oct 15, 2009 at 08:42:22PM +0100, Dan Journo wrote:
Hi Danny,
I've tried that but I get the following errors:-
[r...@templateasteriskserver ~]# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
FATAL: Module dahdi not found.
The dahdi kernel modules are not available. At least not for your
running kernel.
This typically means you have either not installed them ('make install'
in the source directory of dahdi-linux if you install from source), or
installed them and then switched to a different kernel version for some
reason.
What is the output of:
uname -a
find /lib/modules -name dahdi.ko
How have you installed dahdi?
What version of asterisk is it?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] question on SIP and call manager
Here are two ways to address this 1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once 2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt) Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt) CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3 rings) Danny thats good to know for extensions.conf but I am using call files. echo Channel: SIP/CCMMAIN/5551212 /tmp/call echo Context: smvoice-test /tmp/call Can I do the Channel: SIP/CCMMAIN/5551212SIP/CCMSLAVE/5551212 in the Channel for the call file? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Callpickup works for outside calls but not inside calls
On Wed, 2009-10-14 at 22:56 -0400, John A. Sullivan III wrote:
Hello, all. I've got a problem where we set up call pickup for a
customer. If the Bob's extension rings and Bob is in Jim's office, Bob
can press the button on his Snom 320 that says Bob and pick up his
line. It works great for calls coming in from the outside but does not
work for internal calls. Internal calls generate a
app_directed_pickup.c:204 pickup_exec: No target channel found for 617
error.
I see an old bug about this where the contexts were not consistent but
ours appear to be consistent. Here are examples of pertinent parts of
the dialplan:
[a10base]
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
; Terry Keeley
; We put these in a10base rather than a10 or a10pub
;so that the spare stations can access them but public cannot
exten = 612,hint,SIP/tkeeley
; Joe Intrabartola
exten = 613,hint,SIP/jintrabartola
; Maryann Lapolla
exten = 614,hint,SIP/mlapolla
; Michael Intrabartola
exten = 616,hint,SIP/mintrabartola
; Vinny De Marco
exten = 617,hint,SIP/vdemarco
; Reception - the Reception desk may ring when someone dials zero
exten = 621,hint,SIP/reception-a10
; Steve McClain
exten = 624,hint,SIP/smcclain
; Amityville Intercom
;exten = 686,1,Dial(SIP/avilleextdoor-a10,60)
;exten = 686,n,Hangup()
exten = _*8XXX,1,Pickup(${EXTEN:2...@a10pub) ; Enable call pickup for hinted
stations
exten = 7998,1,VoiceMailMain(${CALLERID(num)}...@a10) ; Direct mail retrieval
exten = 7998,n,Hangup()
include = a10pub
include = a10utils
include = a10conf
include = a10parking
[a10in] ; direct inbound SIP dialing
exten = conference,1,Goto(a10pub,6000,1)
exten = joe,1,Goto(a10pub,613,1)
exten = maryann,1,Goto(a10pub,614,1)
exten = michael,1,Goto(a10pub,616,1)
exten = terry,1,Goto(a10pub,612,1)
exten = tommyvan,1,Goto(a10pub,615,1)
exten = vinny,1,Goto(a10pub,617,1)
exten = ebc,1,Goto(a10pub,9,ringall)
exten = vmail,1,Goto(a10pub,7999,1)
[a10pub]
; Public access - BE SURE there is no outbound access from here, e.g.,
; Background() functions will jump to any valid extension entered
; whether or not it is listed in the menu
; Terry Keeley
exten = 612,1,Set(__VM=612) ; VoiceMail ID
exten = 612,n,Gosub(a10ringtones,internal,1)
exten = 612,n,Macro(common,SIP/tkeeley,1,a10)
; 1 for VM, a10 VM context, no followme, ring for default seconds
exten = 8612,1,VoiceMail(6...@a10,u)
exten = 7612,1,VoiceMailMain(6...@a10)
exten = 7612,n,Hangup()
; Joe Intrabartola
exten = 613,1,Set(__VM=613)
exten = 613,n,Gosub(a10ringtones,internal,1)
exten = 613,n,Macro(common,SIP/jintrabartola,1,a10)
exten = 8613,1,VoiceMail(6...@a10,u)
exten = 7613,1,VoiceMailMain(6...@a10)
; Vinny De Marco
exten = 617,1,Set(__VM=617)
exten = 617,n,Gosub(a10ringtones,internal,1)
exten = 617,n,Macro(common,SIP/vdemarco,1,a10)
exten = 8617,1,VoiceMail(6...@a10,u)
exten = 7617,1,VoiceMailMain(6...@a10)
; Floral Park Spare
exten = 618,1,Gosub(a10ringtones,internal,1)
exten = 618,n,Dial(SIP/sparef1-a10,120,o) ; Ring the phone for up to 2
minutes
exten = 618,n,Hangup()
If I make a SIP call across the Internet to Vinny, for example, we issue
a goto to Vinny's internal extension. Terry can press the call pickup
and it all works. The same if I dial in from the PSTN. Here is the
call sequence:
-- Executing [vi...@a10in:1] Goto(SIP/jasiii-ad0e1048, a10pub,617,1)
in new stack
-- Goto (a10pub,617,1)
-- Executing [...@a10pub:1] Set(SIP/jasiii-ad0e1048, __VM=617) in new
stack
-- Executing [...@a10pub:2] Gosub(SIP/jasiii-ad0e1048,
a10ringtones,internal,1) in new stack
-- Executing [inter...@a10ringtones:1]
SIPAddHeader(SIP/jasiii-ad0e1048, Alert-Info:
http://www.notused.com\;info=alert-internal\;x-line-id=0) in new stack
-- Executing [inter...@a10ringtones:2] Return(SIP/jasiii-ad0e1048, )
in new stack
-- Executing [...@a10pub:3] Macro(SIP/jasiii-ad0e1048,
common,SIP/vdemarco,1,a10) in new stack
-- Executing [...@macro-common:1] Set(SIP/jasiii-ad0e1048, TM=24) in
new stack
-- Executing [...@macro-common:2] Dial(SIP/jasiii-ad0e1048,
SIP/vdemarco,24,o) in new stack
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
-- Called vdemarco
-- SIP/vdemarco-d4012df8 is ringing
-- SIP/vdemarco-d4012df8 is ringing
-- SIP/vdemarco-d4012df8 is ringing
-- SIP/vdemarco-d4012df8 is ringing
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
== Extension Changed 612[a10base] new state InUse for Notify User
jintrabartola
== Extension Changed 612[a10base] new state InUse for Notify User
reception-a10
-- Executing [*8...@a10f:1] Pickup(SIP/tkeeley-acc9aaf8, 6...@a10pub)
in new stack
== Extension
[asterisk-users] Mixing SIP/TDM in MeetMe
I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to* the conference. It can hear the conference and its indicator changes appropriated from not talking to talking, but nothing from it gets bridged into the conference (the entering and leaving tones work fine). Calls from the SIP phone to a TDM are fine. I tried the experiment of having the SIP phone dial across the T1 to the PBX which will then tandem the call back to Asterisk. When I do that, I have sound just fine. core show channel look the same for both the Dahdi and SIP channels. This is very frustrating. Does anybody have any ideas? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] The City of Amsterdam has been deploying asterisk throughout the city!
Hi, As you may know by now, yesterday on the Astricon the City of Amsterdam presented their large scale asterisk deployment of 2 phones. Because they do not allow brand names to be used within the city, they call it 'IP Business Manager', but the software they use is in fact the Astium PBX, by NeoNova. Since we are very proud of this project, we have made the Astium available for download from the NeoNova website. You will need a pristine CentOS machine, but it installs easily, and even though it is written for the European market you may want to have a look.. If you're interested, here is the press release: http://www.neonova.nl/nl/content/press/?tid=129735 Thanks, Ron Arts CEO, NeoNova, The Netherlands. -- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Science Park 140 1098 XG Amsterdam info: 020-5611300 servicedesk: 020-5611302 fax: 020-5611301 KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
