Re: [asterisk-users] AMI able to call from known endpoint to unknown endpoint?
On Thu, Jul 31, 2008 at 05:28:42PM -0700, Stephen Cattaneo wrote: Both are sitting behind a Linksys IP PBX (SPA9000). On the Linksys IP PBX I have set the outside number 5000 to connect to 3001. 3002 does not have a similar external mapping (this would defeat the purpose of the test I am attempting). [...] Is it possible (and if yes, how can I do this) to use the AMI's originate to call from 5000 to 3001? Don't you mean to 3002 ? either that or you will make a loopback call ... Anyway, if that's what i understood it's impossible. You can see a PABX a little bit little a network NAT device (well in your specific problem). Behind the Linksys there is a private network, with one public adresse, which is a static map to one private adresse. Well at least it's how your asterisk IPBX will see things. Basically when using asterisk to connect (call) two extension, asterisk will dial each extension, and then connect them. Since he doesn't have any access to your 3002 internal extension, there is no way this could work. When calling from 3001 you can reach the 3002 extension, but only because you are within the same network, to keep the metaphor -- Benoit ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dutch sound files
Hi there, Is there anyone on this list that still has the Dutch sound files for asterisk? When I search for them, I can only find them at http://www.borndigital.nl/voip.asp?page=106title=Asterisk_Soundfiles But the download link doesn't work anymore. Any help would be appriciated. Regards, Roelof Dijkstra Infrastructuremanagement Menzis Zorg en Inkomen, location Enschede ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whitepaper: How and to whom sell VoIP
As for me i mostly saw spellings mistakes, but that's me :) Grygoriy Dobrovolskyy a écrit : i saw that billing iface somewhere else, maybe i am wrong... 2008/7/30 Mindaugas Kezys [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hello, Based on our own and our clients' experience we compiled short manual: How and to whom sell VoIP Hope it can be useful to some of you also. You can download it from our site: http://www.kolmisoft.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Russian Calling card Voice prompt
Dear all Does anyone know where I can find some good quality Russian language voice file for calling card? Thanks in advanced? Sam ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Comparing origination from CLI and from AMI
Hi, Using FOP, I've met a situation which makes me ask this simple question : Are both A and B commands bellow equivalent ? A. CLI: originate SIP/9122 application dial Local/[EMAIL PROTECTED] B. AMI/FOP: 192.168.64.5 - Action: Originate 192.168.64.5 - Channel: SIP/9122 192.168.64.5 - Async: True 192.168.64.5 - Callerid: 9122 Guest2 9122 192.168.64.5 - Exten: 9123 192.168.64.5 - Context: local 192.168.64.5 - Priority: 1 I must add both 9122 and 9123 extensions are SIP extensions which default to local context. When using B (AMI/FOP), I've got a : -- Got SIP response 480 Temporarily Unavailable back from 192.168.100.195 Channel SIP/9122-081d8f68 was never answered. where 192.168.100.195 is SIP/9122 hardphone IP address When using A (CLI), everything works ok. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Authentication on an LDAP server
Hello, I installed Asterisk 1.6.0-beta9 and I desire to authenticate registering users on an LDAP server, before letting them use my Asterisk service. I unfortunately don't succeed in finding enough referrals to achieve my aim. Could you please give me some hints about this topic? Thank you for your help. Ciao Salvo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dutch sound files
On Fri, August 1, 2008 08:59, Dijkstra, Roelof wrote: Hi there, Is there anyone on this list that still has the Dutch sound files for asterisk? When I search for them, I can only find them at http://www.borndigital.nl/voip.asp?page=106title=Asterisk_Soundfiles But the download link doesn't work anymore. Any help would be appriciated. Hey Roelof, I put them online for you, here's the link: http://www.djslash.org/upload/asterisksounds-0.4.tar.gz Have fun! Kind regards, Rutger -- Ytec E-commerce and Business Intelligence M: [EMAIL PROTECTED] T: 0031 502103580 W: www.ytec.nl Onze Algemene Voorwaarden zijn ten alle tijden van toepassing. Deze zijn te vinden op http://www.ytec.nl/av.pdf Op verzoek sturen wij u een geprinte versie toe. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dutch sound files
Hi Rutger, Thanks! Got them. Regards, Roelof -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Rutger van Sleen Verzonden: vrijdag 1 augustus 2008 12:03 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Dutch sound files On Fri, August 1, 2008 08:59, Dijkstra, Roelof wrote: Hi there, Is there anyone on this list that still has the Dutch sound files for asterisk? When I search for them, I can only find them at http://www.borndigital.nl/voip.asp?page=106title=Asterisk_Soundfiles But the download link doesn't work anymore. Any help would be appriciated. Hey Roelof, I put them online for you, here's the link: http://www.djslash.org/upload/asterisksounds-0.4.tar.gz Have fun! Kind regards, Rutger -- Ytec E-commerce and Business Intelligence M: [EMAIL PROTECTED] T: 0031 502103580 W: www.ytec.nl Onze Algemene Voorwaarden zijn ten alle tijden van toepassing. Deze zijn te vinden op http://www.ytec.nl/av.pdf Op verzoek sturen wij u een geprinte versie toe. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queues problem
Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue myqueue In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Filename for Incoming Agent Calls
Hi Ricardo,
Try this:
exten = s,11,Set(MONITOR_FILENAME=/var/spool/queues/PA-${UNIQUEID})
exten = s,12,Set(TRANSFER_CONTEXT=queuetransfer)
exten = s,13,queue(q-pa|t|||)
The TRANSFER_CONTEXT is used for transfers. If you need the filename
inherited, add a double underscore before it.
Thanks
l.
In data Wed, 30 Jul 2008 23:09:11 +0200, Ricardo Melendez
[EMAIL PROTECTED] ha scritto:
Hi, to all, I have configured 3 Inbound/outbound agents queues, I
record
Outgoing calls with custom filename like
outgoing-${callerid(num)}-${EXTEN}-${TIMESTAMP}.gsm
but I need to record Incoming calls and asterisk by default add 13 digits
number to inbound recordings like Agent-001-1298375678-890.gsm, how I
can
customize this filename recordings?
Thanks in advance.
Ricardo Melendez
--
Home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] comparing pots solutions for asterisk
Eric Fort wrote: I've been looking at various solutions for getting FXS and FXO lines in and out of asterisk. one solution is using TDM-400 cards. Another solution is using the grandstream GXW400x and GXW410x gateways. Cost per port seems lower on the gateways and no pci slot is required. Why would one choose to use the TDM-400 cards? what would be the advantages and disadvantages of each approach? The internal card should give you higher reliability as there are fewer parts and cables although the external gateways could allow you to have redundant servers. External gateways would also be easier to scale when you need more lines. Does anyone have experience with the Grandstream gateways? Are they more reliable than their GXP2000 phones? Can they actually provide support yet? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk appliance A50 vs asterisk open source + fxo cards
Hi list, I need advice on which solution to implement, asterisk appliance A50 or just install linux on a pc and get tdm cards. Any comments? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Failover
Tzafrir Cohen wrote: On Thu, Jul 31, 2008 at 02:10:34PM -0400, Dave Welsh wrote: If I buy two AA50s can I set them up so that everything runs through the first one, but the second one will take over if the first one goes down? I can see the extensions recovering, because they use ethernet, but what about the FSO lines? Is there a way they can be spliced to both AA50s so that no one need to do any emergency rewiring? Thinking aloud: what happens if you just connect th two units on the same phone line? Use one of those fax/modem/phone sharing boxes that blocks the other devices when one is off hook. You basically need a way to prevent the slave unit from answering calls if the master is alive. Set the primary unit to pick up on, say, 2 rings and the secondary unit to pick up on 4. If the primary fails, it won't pick up the line and the secondary takes over. You could try having your phones register to both but I'm not sure how that would work regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HI ~ good friend,
hi ~ nice to meet you, i just join here, today, i am a student, and i am very interesting in asterisk. and i have a IP-PBX server, made by me with my friend, while when i studying, i have a question, is there any limit users for asterisk? ex) registed users number is 1000 or 1 or 10 like that, is that possible? and how about the concurrent calls? 1000 concurrent calls is possible? or 2000 concurrent calls? my PBX server's user is just less then 15, almost my friends, so, i can't test, over 10 users and 1000 concurrent calls, please tell me, it is possible or not? thanks your permission to join there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues problem
Hi, I was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got eve nt 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch * Hungup 'Zap/17-1' Kindly give me a hint abt what is happening. And also why my agents are not getting in the queues. Thanks for quick reply. Syed nasr From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Friday, August 01, 2008 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Queues problem Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue myqueue In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues problem
Syed Nasruddin wrote: Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue “myqueue”_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It’s a production machine. Thanks Syed nasr When diagnosing this sort of issue, it is a good idea to check the value of QUEUESTATUS to see why the caller could not enter the queue. The most common reason for a caller to not join the queue is because joinempty=no is set in queues.conf (if you do not have joinempty set at all, then it defaults to no). This setting causes callers attempting to join a queue to not be able to if the queue is empty or if all the queue members are paused or have an invalid device state. Another possibility is that you have a maximum length set on the queue and so no more callers can join because the queue is full. My suggestion is to see what the QUEUESTATUS is. If the status is JOINEMPTY, then you can issue a queue show command on the CLI to see what the current states of your queue members are. It may be as easy to fix as setting joinempty=yes in queues.conf. If the status is something else, though, then a different fix may be in order instead. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Failover
People hate to wait, as a fact i consider AA50s no suited for redundancy, and i am not sure that they were made for that. 2008/8/1 Drew Gibson [EMAIL PROTECTED] Tzafrir Cohen wrote: On Thu, Jul 31, 2008 at 02:10:34PM -0400, Dave Welsh wrote: If I buy two AA50s can I set them up so that everything runs through the first one, but the second one will take over if the first one goes down? I can see the extensions recovering, because they use ethernet, but what about the FSO lines? Is there a way they can be spliced to both AA50s so that no one need to do any emergency rewiring? Thinking aloud: what happens if you just connect th two units on the same phone line? Use one of those fax/modem/phone sharing boxes that blocks the other devices when one is off hook. You basically need a way to prevent the slave unit from answering calls if the master is alive. Set the primary unit to pick up on, say, 2 rings and the secondary unit to pick up on 4. If the primary fails, it won't pick up the line and the secondary takes over. You could try having your phones register to both but I'm not sure how that would work regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk appliance A50 vs asterisk open source + fxo cards
As Drew Gibson wrote at anotehr topic: [quote] The internal card should give you higher reliability as there are fewer parts and cables although the external gateways could allow you to have redundant servers. External gateways would also be easier to scale when you need more lines. [/quote] 2008/8/1 James Mutuku [EMAIL PROTECTED] Hi list, I need advice on which solution to implement, asterisk appliance A50 or just install linux on a pc and get tdm cards. Any comments? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues problem
Syed Nasruddin a écrit : Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue “myqueue”_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It’s a production machine. Thanks Syed nasr I would recommend upgrading your asterisk to at least 14.20.1 I have had many troubles with queues, SIP and IAX with asterisk 1.4.18 that have been fixed in the following releases ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Failover
Set the primary unit to pick up on, say, 2 rings and the secondary unit to pick up on 4. If the primary fails, it won't pick up the line and the secondary takes over. You could try having your phones register to both but I'm not sure how that would work regards, Drew That's exactly how I have things setup. Phones register to both servers. The backup server doesn't do much - only answers incoming calls if the ring goes on for too long. Keep in mind that you may want to configure the backup server to not use the lines for placing outbound calls - maybe setup a SIP trunk for that. Otherwise, when the primary server comes back up, the two will be talking over each other trying to use the same lines. --Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Failover
On Fri, 1 Aug 2008 09:20:07 -0500, Brent Torrenga wrote: Set the primary unit to pick up on, say, 2 rings and the secondary unit to pick up on 4. If the primary fails, it won't pick up the line and the secondary takes over. You could try having your phones register to both but I'm not sure how that would work regards, Drew That's exactly how I have things setup. Phones register to both servers. The backup server doesn't do much - only answers incoming calls if the ring goes on for too long. Keep in mind that you may want to configure the backup server to not use the lines for placing outbound calls - maybe setup a SIP trunk for that. Otherwise, when the primary server comes back up, the two will be talking over each other trying to use the same lines. Polycom phones support primary and secondary servers. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HI ~ good friend,
Hi welcome to the asterisk community. The answer you want are here; http://www.voip-info.org/wiki/view/Asterisk+dimensioning The short answer is; Pretty much yes, depending on hardware and horizontal scaling with multiple servers sharing the load. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ??? Sent: Friday, 1 August 2008 9:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] HI ~ good friend, hi ~ nice to meet you, i just join here, today, i am a student, and i am very interesting in asterisk. and i have a IP-PBX server, made by me with my friend, while when i studying, i have a question, is there any limit users for asterisk? ex) registed users number is 1000 or 1 or 10 like that, is that possible? and how about the concurrent calls? 1000 concurrent calls is possible? or 2000 concurrent calls? my PBX server's user is just less then 15, almost my friends, so, i can't test, over 10 users and 1000 concurrent calls, please tell me, it is possible or not? thanks your permission to join there, http://mail.nate.com/web/footer/img/jem.gif http://mail.nate.com/web/footer/img/logo_nate_20060420.gif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues problem
Thanks, Yes that was the problem I have added joinempty=yes. It is now working,. Right now another critical problem has come up which I have mentioned in my previous email. I am copying the problem here again: was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got event 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch Hungup 'Zap/17-1' Please help on this urgent. I cant upgrade right now since I am not confident abt upgrade procedure and any other problems occuring after that. This is my only production machine. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Friday, August 01, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Syed Nasruddin wrote: Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue myqueue_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr When diagnosing this sort of issue, it is a good idea to check the value of QUEUESTATUS to see why the caller could not enter the queue. The most common reason for a caller to not join the queue is because joinempty=no is set in queues.conf (if you do not have joinempty set at all, then it defaults to no). This setting causes callers attempting to join a queue to not be able to if the queue is empty or if all the queue members are paused or have an invalid device state. Another possibility is that you have a maximum length set on the queue and so no more callers can join because the queue is full. My suggestion is to see what the QUEUESTATUS is. If the status is JOINEMPTY, then you can issue a queue show command on the CLI to see what the current states of your queue members are. It may be as easy to fix as setting joinempty=yes in queues.conf. If the status is something else, though, then a different fix may be in order instead. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic Door phone monitor to Asterisk box?
the Panasonic model you mention will only work with a panasonic PBX. and no it does NOT plug into a station port on the Panasonic but onto the door card of any of the Panasonic systems. Having installed that Panasonic video doorboxes in the past I would suggest stay away from it. it is not as nice as the brouchures make it look, and extremly overpriced. On 7/31/08, Steve Prior [EMAIL PROTECTED] wrote: I'm considering getting a Panasonic video door phone system (VL-GM301A) which can interface with a PBX and would like to connect it to my Asterisk box with an analog FXS port. Of course the Panasonic documentation only talks about hooking it up to a Panasonic PBX which only talks about using Panasonic phones, so it's hard to tell whether the 2 wire connection from the door phone monitor is analog or digital. The video door phone itself hooks to the door phone monitor with a 2 conductor wire, so that part is clearly digital since video, audio, and button press all go through that wire, but since the door phone central station apparently plugs into a Panasonic PBX FXS port which possibly supports fax machines I'm thinking that this might be an ordinary analog connection. Does anyone know if the door phone interface is analog or digital? Anyone have experience with interfacing with one? Thanks Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic Door phone monitor to Asterisk box?
C F wrote: the Panasonic model you mention will only work with a panasonic PBX. and no it does NOT plug into a station port on the Panasonic but onto the door card of any of the Panasonic systems. Having installed that Panasonic video doorboxes in the past I would suggest stay away from it. it is not as nice as the brouchures make it look, and extremly overpriced. Glad I asked. Here's what I REALLY want to do... I like the look of the Panasonic door unit - looks like a simple doorbell, but has a camera, mic, speaker in there as well as the button. I want the camera (NTSC) hooked to a video capture card full time so that system can always see outside. I'm happy with a plain old doorbell sounding when the button is pressed. I don't care about the inside wall mounted panel at all, and I'd be happy to have the mic/speaker connected to my asterisk box somehow. This means: - the doorbell works with a chime - my home automation/security system can always view out the camera - I can call the outside unit from my Asterisk PBX and talk to whoever is out there. It all starts with the outside hardware, do you (or anyone else) know of a nice looking unit preferably along the lines of the look of the Panasonic unit? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Failover
Brent Torrenga wrote: Set the primary unit to pick up on, say, 2 rings and the secondary unit to pick up on 4. If the primary fails, it won't pick up the line and the secondary takes over. You could try having your phones register to both but I'm not sure how that would work regards, Drew That's exactly how I have things setup. Phones register to both servers. The backup server doesn't do much - only answers incoming calls if the ring goes on for too long. Keep in mind that you may want to configure the backup server to not use the lines for placing outbound calls - maybe setup a SIP trunk for that. Otherwise, when the primary server comes back up, the two will be talking over each other trying to use the same lines. --Brent That's where the fax/modem sharing box comes in, if one device port has the line, all other device ports are blocked. Downside is that this will also cause dead air on an outbound call or two until things settle down again. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto provisioning phones
Which Asterisk systems provide automatic provisioning of phones? Switchvox? ABE? The AA series appliances? Trixbox? I know that the VDEX-40 (Voiceroute) and Jazinga do this. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto provisioning phones
Michael Graves wrote: Switchvox? ABE? The AA series appliances? All three of these can auto-provision Polycom phones, but that is the only brand that they support for provisioning. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto provisioning phones
Michael, Please take a look the Druid, Open Source Unified Communications http://www.voiceroute.org We have auto-provisioning support for Cisco, Polycom, Snom, Mitel, Aastra, Grandstream, Linksys Ming On Fri, Aug 1, 2008 at 11:36 PM, Michael Graves [EMAIL PROTECTED] wrote: Which Asterisk systems provide automatic provisioning of phones? Switchvox? ABE? The AA series appliances? Trixbox? I know that the VDEX-40 (Voiceroute) and Jazinga do this. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- Meet us at LinuxWorld 2008, 4-7 Aug 2008, Moscone Center, San Francisco, Booth 1626 http://druidlinuxworld.eventbrite.com Meet us at WEB 2.0 EXPO, 16-19 Sept 2008, Javits Center, NYC, Booth #17 http://druidweb20.eventbrite.com See Voiceroute OSCON 2008 Druid project presentation on youtube http://www.youtube.com/watch?v=2gfIAXm5vTc http://www.youtube.com/watch?v=dkm6P4O0oac ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Dundi] Looking for new peers/limited time only
On Thursday 31 July 2008 11:36:18 am Anthony Messina wrote: For a limited time only, Messinet Secure Services (me) will be offering DUNDi E.164 termination to the entire +1 country code. I'd like to encourage more peering within the US, but peering is open to anyone. See http://messinet.com/?page_name=DUNDi for peering information. Between 11AM yesterday and 11AM today, Messinet Secure Services serviced over 430 free calls totaling over 18 hour and 15 minutes of free calling time to any number using the +1 country code. All this was accomplished via DUNDi E.164 peering. Judging by the numbers for yesterday, today will probably be the last day this offer, so peer up now! See http://messinet.com/?page_name=DUNDi for peering information. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic Door phone monitor to Asterisk box?
On Fri, Aug 1, 2008 at 11:09 AM, Steve Prior [EMAIL PROTECTED] wrote: C F wrote: the Panasonic model you mention will only work with a panasonic PBX. and no it does NOT plug into a station port on the Panasonic but onto the door card of any of the Panasonic systems. Having installed that Panasonic video doorboxes in the past I would suggest stay away from it. it is not as nice as the brouchures make it look, and extremly overpriced. Glad I asked. Here's what I REALLY want to do... I like the look of the Panasonic door unit - looks like a simple doorbell, but has a camera, mic, speaker in there as well as the button. I want the camera (NTSC) hooked to a video capture card full time so that system can always see outside. I'm happy with a plain old doorbell sounding when the button is pressed. I don't care about the inside wall mounted panel at all, and I'd be happy to have the mic/speaker connected to my asterisk box somehow. This means: - the doorbell works with a chime I beleive Aiphone AX will do that. - my home automation/security system can always view out the camera Again the Aiphone AX should do that. BTW, the Panasonic one turns off after 1 minute. There is no way to keep it on all the time AFAIK. - I can call the outside unit from my Asterisk PBX and talk to whoever is out there. I think that the Aiphone AX will do that. Not 100% sure though. Here is the instructions and Tech Support is very good: http://www.aiphone.com/PDF_Files/Prod_Lit/Stock_InstrInstal/AX-Manual%20(low%20Res).pdf In most cases such a setup is cheaper and better if you do separate camera and door box, using some multiplexer that makes sure screen switches to that door camera on call button. It all starts with the outside hardware, do you (or anyone else) know of a nice looking unit preferably along the lines of the look of the Panasonic unit? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up ring group
Tom Moore wrote:
Hi guys,
What's the best way to setup a ring group that contains 6 extensions so that
when a call comes in there starts a 30 second timer and the first available
device is rang instead of ringing all extensions at the same time?
What I want it to do is cycle through the extensions and have the system
ignore the ones that are busy and if there are not any free extensions in
the ring group to have the system drop the caller to voicemail.
If none of the extensions are present in the group I'd like to also drop to
voicemail.
Basically what I'm looking for is a multiple extensions version of the
standard extension macro with multiple devices and the exten busy state
ignored.
Tom
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Hi I had the same problem. At the beginning I thought of implementing
agents and queues. But that's not what I wanted. I didn't go on and look
how to configure members (perhaps that would've been the better
solution), maybe because I'm always thinking on how to program something
and I'm not always aware that there are already solutions to many
problems out there.
Anyway, that's how it looks like in my extensions.conf
[wait-op]
; Ask if the channel is available, if it is
; go to the next step. If it isn't go to no-op
; and skip the delay.
exten = _XX,1,ChanIsAvail(SIP/${EXTEN})
exten = _XX,n,GotoIf($[ ${AVAILCHAN}= ]?no-op|s-na|1:3)
; Increment the delay by a value of five.
exten = _XX,n,Set(DB(cross/delay-${key})=$[${DB(cross/delay-${key})}+5])
exten = _XX,n,Wait(${DB(cross/delay-${key})})
exten = _XX,n,Dial(SIP/${EXTEN})
[no-op]
; Do nothing
exten = s,1,NoOp(Dummy)
exten = s-na,1,NoOp(Channel is not available)
[hotline-0]
; Define a custom name for the caller ID.
; This was an extra that I did
exten = s,1,Set(CALLERID(name)=hotline ${CALLERID(name)} ${CALLERID(num)})
; Set a key unique for each channel. So id doesn't matter how
; many calls we get, there will always exist just one key per channel
; This way we increase the delay only when we want to.
exten = s,n,Set(__key=${CHANNEL})
; Define the initial delay value on the database. That's even better than
; a global variable. One advantage, pointed out by a collegue of mine, is
; that when the process is over, you can delete the key from the DB.
exten = s,n,Set(DB(cross/delay-${key})=-5)
; Set all the devices as a single variable.
; Note that all of them use the Local context
exten = s,n,Set(dg0=Local/[EMAIL PROTECTED])
exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED])
exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED])
exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED])
exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED])
exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED])
exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED])
exten = s,n,Dial(${dg0}|80)
; Manage the voicemail with a macro
exten = s,n,Macro(hotline-voicemail|${DIALSTATUS}|0)
; Delete the keys at hangup
exten = h,1,NoOp(DB_DELETE(cross/inc-${key})
exten = h,n,Hangup
[macro-hotline-voicemail]
; ${ARG1} Dialstatus
; ${ARG2} Whose voicemail?
exten = s,1,Set(CHANNEL(language)=de)
exten = s,n,Goto(s-${ARG1},1)
exten = s-BUSY,1,Voicemail(${ARG2},b)
exten = s-NOANSWER,1,Voicemail(${ARG2},u)
exten = s-CONGESTION,1,Voicemail(${ARG2},b)
exten = s-CHANUNAVAIL,1,Voicemail(${ARG2},u)
[default]
exten = 0,1,Goto(hotline-0|s|1)
...
I hope it works for you :)
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[asterisk-users] how many quad T1 cards
Assuming you have a Quad core machine, at least 4 GIG ram, will a machine like this handle 4 Quad T1 cards? is that advisable? What about running AGI's on such a machine. Will the machine handle starting/stopping all those AGI's? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_flite 0.6 released
I've updated the app_flite module to work with the Asterisk 1.6.x code- base in addition to it already working with the 1.4.x, and 1.2.x. (1.0.x support is untested and unsupported). It can be downloaded on my website at: http://www.darrensessions.com/downloads/app_flite-0.6.tar.gz Additional details are in the ChangeLog and README files in the tar ball. As always, if there are any questions or comments, please forward them to me at [EMAIL PROTECTED] Thanks, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Logs
I'm impressed with the call log reporting that Switchvox offers. There's a screenshot here: http://www.switchvox.com/sv?cmd=screenshotspic=13 Can normal Asterisk do this? I installed AsteriskNow on an old computer, but I couldn't find anything like that in the GUI. I found the CSV file in /var/log/asterisk/cdr-csv, but that's not much help unless there's some other software that can turn the CSV into something more user friendly. Is there software like that? -- Dave Welsh Quality of Course (613) 749-8248 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Logs
Dave, Druid open source edition (http://www.voiceroute.org) has something similar where we actually parse through the CDR logs to present the call flow Check it out http://www.voiceroute.org/druidose/screenshots/call_records Ming On Sat, Aug 2, 2008 at 2:34 AM, Dave Welsh [EMAIL PROTECTED] wrote: I'm impressed with the call log reporting that Switchvox offers. There's a screenshot here: http://www.switchvox.com/sv?cmd=screenshotspic=13 Can normal Asterisk do this? I installed AsteriskNow on an old computer, but I couldn't find anything like that in the GUI. I found the CSV file in /var/log/asterisk/cdr-csv, but that's not much help unless there's some other software that can turn the CSV into something more user friendly. Is there software like that? -- Dave Welsh Quality of Course (613) 749-8248 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- Meet us at LinuxWorld 2008, 4-7 Aug 2008, Moscone Center, San Francisco, Booth 1626 http://druidlinuxworld.eventbrite.com Meet us at WEB 2.0 EXPO, 16-19 Sept 2008, Javits Center, NYC, Booth #17 http://druidweb20.eventbrite.com See Voiceroute OSCON 2008 Druid project presentation on youtube http://www.youtube.com/watch?v=2gfIAXm5vTc http://www.youtube.com/watch?v=dkm6P4O0oac ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
You mean running , 400 Calls on 1 BOX ? Even if you COULD do it, the gods of TELCO would have you burn in hell for stacking that much critical traffic on ONE Intel, non - high availability box Jerry Geis wrote: Assuming you have a Quad core machine, at least 4 GIG ram, will a machine like this handle 4 Quad T1 cards? is that advisable? What about running AGI's on such a machine. Will the machine handle starting/stopping all those AGI's? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Logs
Hey, What you're looking at in that gui is just an organized representation of the cdr generated by asterisk. There are some tools available that do this, you can look at asterisk-stat: http://www.voip-info.org/tiki-index.php?page=Asterisk+CDR+Areski+GUI I can't think of any others right now but potentially you could write your own fronted by having it pull the info from the cdr. Regards, Igor H. Dave Welsh wrote: I'm impressed with the call log reporting that Switchvox offers. There's a screenshot here: http://www.switchvox.com/sv?cmd=screenshotspic=13 Can normal Asterisk do this? I installed AsteriskNow on an old computer, but I couldn't find anything like that in the GUI. I found the CSV file in /var/log/asterisk/cdr-csv, but that's not much help unless there's some other software that can turn the CSV into something more user friendly. Is there software like that? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
On Friday 01 August 2008 13:20:55 Jerry Geis wrote: Assuming you have a Quad core machine, at least 4 GIG ram, will a machine like this handle 4 Quad T1 cards? is that advisable? What about running AGI's on such a machine. Will the machine handle starting/stopping all those AGI's? I'm not terribly sure that the PCI bus will stand up to that many interrupts per second, though it's certainly possible. Last I heard the PCI bus was nearly at capacity servicing just 3 quad-span cards (note that the PCI bus has other things to service, like hard drive accesses, network, keyboard, etc.). You'll probably do better with two machines, rather than trying to stack everything into one. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip show peer [load] says not a realtime peer
When I do a sip show peer peer load command in the Asterisk CLI I get the information about the peer I requested, however, there is a line that says Realtime peer: No. All the other information is correct. According to help sip show peer the Option load forces lookup of peer in realtime storage.. Also, this particular peer is only defined in the database (I greped all the files in the /etc/asterisk directory to make sure). Why is my realtime peer not being reported as a realtime peer? I ran across the above command while trying to figure out why my realtime peers (which register just fine) are not shown with the sip show peers command. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Filename for Incoming Agent Calls
Could you clarify what you mean as inherited
In The dial plan for a given call I thought All variable were GLOBAL
to that call ??
Thanks
lenz wrote:
Hi Ricardo,
Try this:
exten = s,11,Set(MONITOR_FILENAME=/var/spool/queues/PA-${UNIQUEID})
exten = s,12,Set(TRANSFER_CONTEXT=queuetransfer)
exten = s,13,queue(q-pa|t|||)
The TRANSFER_CONTEXT is used for transfers. If you need the filename
inherited, add a double underscore before it.
Thanks
l.
In data Wed, 30 Jul 2008 23:09:11 +0200, Ricardo Melendez
[EMAIL PROTECTED] ha scritto:
Hi, to all, I have configured 3 Inbound/outbound agents queues, I
record
Outgoing calls with custom filename like
outgoing-${callerid(num)}-${EXTEN}-${TIMESTAMP}.gsm
but I need to record Incoming calls and asterisk by default add 13 digits
number to inbound recordings like Agent-001-1298375678-890.gsm, how I
can
customize this filename recordings?
Thanks in advance.
Ricardo Melendez
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[asterisk-users] Scour.com invite from rahul
Hey, Did you hear about Scour? It is the next gen search engine with Google/Yahoo/MSN results and user comments all on one page. Best of all we get paid for using it by earning points with every search, comment and vote. The points are redeemable for Visa gift cards! It's like earning credit card or airline points just for searching! Hit the link below to join for free and we will both get points! http://scour.com/invite/mastrao1/ I know you'll like it! - rahul ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
If you had a dax in front of all your circuits, you could move them from one server to another without physically touching anything. I've done about 300 calls on a dual processor box doing just SIP with an entirely AGI based setup and it held up just fine, but doing TDM, I'd worry about your PCI bus at those call levels. - D _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 1, 2008, at 1:07 PM, Al Baker wrote: You mean running , 400 Calls on 1 BOX ? Even if you COULD do it, the gods of TELCO would have you burn in hell for stacking that much critical traffic on ONE Intel, non - high availability box Jerry Geis wrote: Assuming you have a Quad core machine, at least 4 GIG ram, will a machine like this handle 4 Quad T1 cards? is that advisable? What about running AGI's on such a machine. Will the machine handle starting/stopping all those AGI's? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
2 or 3 cheaper gateway machines that have just the T1 cards in it, will do way better than one monstrous machine. On Fri, Aug 1, 2008 at 2:20 PM, Jerry Geis [EMAIL PROTECTED] wrote: Assuming you have a Quad core machine, at least 4 GIG ram, will a machine like this handle 4 Quad T1 cards? is that advisable? What about running AGI's on such a machine. Will the machine handle starting/stopping all those AGI's? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
On Fri, Aug 01, 2008 at 03:24:51PM -0400, C F wrote: 2 or 3 cheaper gateway machines that have just the T1 cards in it, will do way better than one monstrous machine. Indeed. We run VICIdial here for about 255 agents, and we're doing that on roughly 11 dialler boxes with 2-3 spans and 15-22 fronters per box, and one closer box with 77 seats on it; 4 channelbanks but no T-spans. Our DBMS is a separate Quad-Opteron with 16GB and we have a smattering of other boxes in the datacenter; our current server count is 41. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Filename for Incoming Agent Calls
On Friday 01 August 2008 14:16:42 Al Baker wrote:
lenz wrote:
Hi Ricardo,
Try this:
exten = s,11,Set(MONITOR_FILENAME=/var/spool/queues/PA-${UNIQUEID})
exten = s,12,Set(TRANSFER_CONTEXT=queuetransfer)
exten = s,13,queue(q-pa|t|||)
The TRANSFER_CONTEXT is used for transfers. If you need the filename
inherited, add a double underscore before it.
Could you clarify what you mean as inherited
In The dial plan for a given call I thought All variable were GLOBAL
to that call ??
Thanks
No, variables are global to the CHANNEL. Calls are bridges between two
channels. Variables are not transferred to a dialled (slave) channel, unless
you set up inheritance, as noted above.
--
Tilghman
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Re: [asterisk-users] HI ~ good friend,
I must disagree. Dimensioning of Asterisk is a very sorely lacking area and is one of the main area CISCO and such eats its lunch. There simply no a base of solid metric that allow for true provisioning . Yes, there are INVALUABLE anecdotal reports from people who have been kind, and sharing of their experiences and for which are all very very grateful. BUT That that just is not the same as as solid, vendor based Metrics. Can you imagine calling and asking DISCO, What do I need for 400 calls an their answer is Here please go read these mostly outdated anecdotal reports and call back with your order Sorry. I love *, but this area of it is not where it needs to be. Dean Collins wrote: Hi welcome to the asterisk community. The answer you want are here; http://www.voip-info.org/wiki/view/Asterisk+dimensioning The short answer is; Pretty much yes, depending on hardware and horizontal scaling with multiple servers sharing the load. Cheers, Dean *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *??? *Sent:* Friday, 1 August 2008 9:43 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] HI ~ good friend, hi ~ nice to meet you, i just join here, today, i am a student, and i am very interesting in asterisk. and i have a IP-PBX server, made by me with my friend, while when i studying, i have a question, is there any limit users for asterisk? ex) registed users number is 1000 or 1 or 10 like that, is that possible? and how about the concurrent calls? 1000 concurrent calls is possible? or 2000 concurrent calls? my PBX server's user is just less then 15, almost my friends, so, i can't test, over 10 users and 1000 concurrent calls, please tell me, it is possible or not? thanks your permission to join there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970, CTLSEPmac.tlv
I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian developers*cough*) package, which is apparently not RFC 783 (tftp) compliant with file not found responses. The whopping 18 page RFC states that Error Code should be 0x00,0x01 for file not found errors, but netkit-tftpd returns 0x00,0x00 which is Not defined - causing the phone to ignore it and request the file again a few seconds later. Solution: Switch to any other tftpd. The moment I switched to tftpd-hpa or atftpd, the phone stopped looping, picked up the SEPmac.cnf.xml file, and immediately registered to Asterisk. Hopefully in the future Debian will rename, remove, or fix this package so it is no longer the default tftpd. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HI ~ good friend,
Yep I totally agree with you that documentation is an area digium is dropping the ball. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Baker Sent: Friday, 1 August 2008 3:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HI ~ good friend, I must disagree. Dimensioning of Asterisk is a very sorely lacking area and is one of the main area CISCO and such eats its lunch. There simply no a base of solid metric that allow for true provisioning . Yes, there are INVALUABLE anecdotal reports from people who have been kind, and sharing of their experiences and for which are all very very grateful. BUT That that just is not the same as as solid, vendor based Metrics. Can you imagine calling and asking DISCO, What do I need for 400 calls an their answer is Here please go read these mostly outdated anecdotal reports and call back with your order Sorry. I love *, but this area of it is not where it needs to be. Dean Collins wrote: Hi welcome to the asterisk community. The answer you want are here; http://www.voip-info.org/wiki/view/Asterisk+dimensioning The short answer is; Pretty much yes, depending on hardware and horizontal scaling with multiple servers sharing the load. Cheers, Dean *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *??? *Sent:* Friday, 1 August 2008 9:43 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] HI ~ good friend, hi ~ nice to meet you, i just join here, today, i am a student, and i am very interesting in asterisk. and i have a IP-PBX server, made by me with my friend, while when i studying, i have a question, is there any limit users for asterisk? ex) registed users number is 1000 or 1 or 10 like that, is that possible? and how about the concurrent calls? 1000 concurrent calls is possible? or 2000 concurrent calls? my PBX server's user is just less then 15, almost my friends, so, i can't test, over 10 users and 1000 concurrent calls, please tell me, it is possible or not? thanks your permission to join there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
I am not about to change the camp here but sipx claim that they can do it with external audiocodes gateways, i have no experience with sipx. Whatever you choose you need a good redundancy here, also a cost in involved, lets see: Quote from their site: *Deployment for large enterprise with more than 1,000 users (up to 5,000 or more): * With deployments larger than about 1,000 users the sipXecs system is typically deployed in a distributed way where it's different components run on dedicated hardware that is centrally managed by the sipXecs configuration and management solution. Typical system partitioning would be as follows: 2 servers for redundant call control, 1 separate server for media services like voicemail and auto-attendant. The configuration management system would typically run on the same hardware as the media server or be installed on dedicated hardware. In addition, if larger call center capabilities are required, the ACD call center server that is part of sipXecs can also run on dedicated hardware. Several ACD servers can be run in parallel where different queues are assigned to different servers. Distributed deployments where sipXecs components spread across several server hardware currently need to be manually installed and configured. Therefore, the installation process requires more technical skill and familiarity with the sipXecs solution to succeed. We are working on creating a cluster management system, where such deployments will become as easy as simpler installations. /QUOTE I suppose same conclusions with asterisk installations 1 4xT1 from digium will cost around 2200$x4= 8800$ Audiocodes mediant 16xT1/E1 $26,699.99 But in the case of asterisk multiserver, you need more T1 Quad card's if primary server's fails for fast switch. I am passing the details like: forget about transcoding, ethernet redundancy, power protection. I took prices from http://www.voipsupply.com/ and from digium site, not working for any, just googled and found prices, they can vary from one to another, also i choosed audiocodes as 'worth it's price' you got lot of other choices/prices/brands. whatever soft you choose :one server is NO way to do, separate, optimise, count money ;) also in case of asterisk: openser, heartbeat. 2008/8/1 Jerry Geis [EMAIL PROTECTED] Assuming you have a Quad core machine, at least 4 GIG ram, will a machine like this handle 4 Quad T1 cards? is that advisable? What about running AGI's on such a machine. Will the machine handle starting/stopping all those AGI's? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scour.com invite from rahul
Hello have you heard about Cuil ? it's a free search done by google engineers, also a bonus i am not putting my reference to win money on your back, have a look: http://www.cuil.com/ 2008/8/1 rahul [EMAIL PROTECTED] Hey, Did you hear about Scour? It is the next gen search engine with Google/Yahoo/MSN results and user comments all on one page. Best of all we get paid for using it by earning points with every search, comment and vote. The points are redeemable for Visa gift cards! It's like earning credit card or airline points just for searching! Hit the link below to join for free and we will both get points! http://scour.com/invite/mastrao1/ I know you'll like it! - rahul ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scour.com invite from rahul
Forgot to add: ;) I know you'll like it! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk appliance A50 vs asterisk open source + fxo cards
Hi, Thanks for the response.Would you rather install asterisk or use the appliance? James Grygoriy Dobrovolskyy wrote: As Drew Gibson wrote at anotehr topic: [quote] The internal card should give you higher reliability as there are fewer parts and cables although the external gateways could allow you to have redundant servers. External gateways would also be easier to scale when you need more lines. [/quote] 2008/8/1 James Mutuku [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi list, I need advice on which solution to implement, asterisk appliance A50 or just install linux on a pc and get tdm cards. Any comments? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk appliance A50 vs asterisk open source + fxo cards
If you know how to configure asterisk, go to proper install with gateways. redundancy is easy to manage in this case, no internal hardware, heartbeat for another server, and your are ready to failover ;) And dont forget about the 'boss', they like sometimes fancy web interfaces, and click stuff ;) If you need a small server with no complications, and no critical failover go for appliance. Choice is yours.2008/8/1 James Mutuku [EMAIL PROTECTED] Hi, Thanks for the response.Would you rather install asterisk or use the appliance? James Grygoriy Dobrovolskyy wrote: As Drew Gibson wrote at anotehr topic: [quote] The internal card should give you higher reliability as there are fewer parts and cables although the external gateways could allow you to have redundant servers. External gateways would also be easier to scale when you need more lines. [/quote] 2008/8/1 James Mutuku [EMAIL PROTECTED] Hi list, I need advice on which solution to implement, asterisk appliance A50 or just install linux on a pc and get tdm cards. Any comments? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto provisioning phones
Druid does I believe. Not sure about any others though. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, August 01, 2008 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] auto provisioning phones Which Asterisk systems provide automatic provisioning of phones? Switchvox? ABE? The AA series appliances? Trixbox? I know that the VDEX-40 (Voiceroute) and Jazinga do this. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto provisioning phones
Elastix ? 2008/8/1 Tom Moore [EMAIL PROTECTED] Druid does I believe. Not sure about any others though. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, August 01, 2008 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] auto provisioning phones Which Asterisk systems provide automatic provisioning of phones? Switchvox? ABE? The AA series appliances? Trixbox? I know that the VDEX-40 (Voiceroute) and Jazinga do this. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970, CTLSEPmac.tlv
On 15:02, Fri 01 Aug 08, Jason Parker wrote: I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian developers*cough*) package, which is apparently not RFC 783 (tftp) compliant with file not found responses. The whopping 18 page RFC states that Error Code should be 0x00,0x01 for file not found errors, but netkit-tftpd returns 0x00,0x00 which is Not defined - causing the phone to ignore it and request the file again a few seconds later. Solution: Switch to any other tftpd. The moment I switched to tftpd-hpa or atftpd, the phone stopped looping, picked up the SEPmac.cnf.xml file, and immediately registered to Asterisk. Hopefully in the future Debian will rename, remove, or fix this package so it is no longer the default tftpd. Thanks for the write-up. I tried with the latest 7960 firmware, and it did work with the default debian tftpd (had to install a new VM) For googleable stuff: The default tftpd on OpenBSD works fine ;) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Filename for Incoming Agent Calls
Wow - Thanks a bunch. Likely save me about 12 hours of struggle
Tilghman Lesher wrote:
On Friday 01 August 2008 14:16:42 Al Baker wrote:
lenz wrote:
Hi Ricardo,
Try this:
exten = s,11,Set(MONITOR_FILENAME=/var/spool/queues/PA-${UNIQUEID})
exten = s,12,Set(TRANSFER_CONTEXT=queuetransfer)
exten = s,13,queue(q-pa|t|||)
The TRANSFER_CONTEXT is used for transfers. If you need the filename
inherited, add a double underscore before it.
Could you clarify what you mean as inherited
In The dial plan for a given call I thought All variable were GLOBAL
to that call ??
Thanks
No, variables are global to the CHANNEL. Calls are bridges between two
channels. Variables are not transferred to a dialled (slave) channel, unless
you set up inheritance, as noted above.
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
On 7/31/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Jul 31, 2008 at 05:36:14PM +1000, Lee, John (Sydney) wrote: Yes, I tried all sorts of cables and ended up getting the local contact to complain to NETCOM. An engineer came and swapped the Fast Ethernet to E1 converter. Hmmm. Whose side is Fast Ethernet, and whose side is E1? Are you trying to take the E1 that they've *converted into 100BT* for you and plug it into an E1 port? Since this thread is still going I thought I'd chime in again. With our working CNC setup in Kunming, they provide some kind of router which breaks a single incoming fibre in to both 100BT and an E1 line that plugs in to the Sangoma card. zaptel_hardware output is: pci::04:06.0 wanpipe- 1923:0300 Sangoma Technologies Corp. A101 single-port T1/E1 /etc/asterisk/zapata.conf: ; Sangoma A102 port 1 [slot:6 bus:4 span:1] wanpipe1 switchtype=5ess context=incoming-kunming group=0 signalling=pri_cpe channel =1-15,17-31 One thing that caused issues when setting up for the first time was the fact that dialling out without setting the correct 'caller ID' would yield errors. So, make sure in your dialplan you do this, or outgoing testing may inexplicably fail. A line like: exten = s,n,Set(CALLERID(number)=02222) Also, if you have not set up an incoming context calling in over the analog network will generate an error tone from the network, rather than anything more obvious. In this case somewhere in asterisk's logfiles you can see unknown extension or an error of that sort that appears each time an incoming attempt is made, but there are no other clues. So make sure your incoming contexts are set up! Best of luck. Regards, Walter Stanish Owner / Director Occident Systems (+86 15808 700 801) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
