[asterisk-users] SIP client on a mobile?
Steve Totaro wrote: Just use a SIP client on your phone. Many providers have multiple failover paths for inbound calls. This thread morphed from a nice home phone system into something completely different. Yup. For my situation, DISA is pointless except for road warriors who call all over the world, from anywhere, they can call into the corp system, get dialtone and skip the whole process of expense reports for work related calls. It makes things less complex, not more. Using DISA also means getting a corp caller id, not a mobile. Yes, spoofing provides that. Maybe if you explain your situation and how your plan works, but for me, personally, DISA would be a an added cost and complication. The only purpose I can think of for myself could be accomplished by spoofing caller id. How is that done from a mobile? Sofar that has been my main reason for using DISA - cost is not a real issue. SIP client. Spoof card, yes it is DISA, but you don't have to do anything but use the card. Steve, even if I could get SIP clients for our phones, doesn't this mean using a data connection rather than just voice? That would make it a lot pricier than the current setup with DISA (which is largely free). /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Steve Totaro wrote: VoIP mostly aside, a couple more thoughts. I am not sure I understand your reasoning for DISA or how it is cheaper. The only reason we use DISA is to spoof the caller id. The OP also wanted to save costs, which is also possible (as someone already confirmed). DISA does save some cost for me too, but it is immaterial. The call from the mobile to the asterisk box is free or flat fee due to calling groups offered by our provider. The outgoing call is charged at regular fixnet prices, much cheaper than mobile ditto. You can buy a card that accepts SIMs as FXO and FXS. For your reasoning, a card of such nature is required. Populate it with different SIMs or whatever that are in calling groups or whatever you were trying to say. You've lost me, I have no idea what you're talking about. Just use callback back and some logic to reduce your costs. Call back will allow you to use the corp identity, and LCR will cut costs over DISA. The system calls you back after you make a call. Then the call is placed. There is a very brief outbound cell phone call, followed by a an inbound call from the server that you initiated with call back. OK, I see. I haven't looked at that, but it sounds more complicated than using DISA, and I'm not convinced it would be any cheaper. (it's important that the scheme be easy to use from the mobile end). Inbound to a cell is generally less expensive that oubound on a cell, sometimes completely free. Yes, inbound to a mobile is free as long as you're not roaming. However, with our calling group setup, it doesn't matter who (fix or mobile) originates the call, the cost is the same. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Steve Totaro wrote: So in other worlds you had nothing to contribute to this thread. I did - you didn't understand my reasoning, I explained it. If you had nothing to contribute to this thread, perhaps you should have stayed away too. /Per Jessen, Zürich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client on a mobile?
Steve Totaro wrote: Steve, even if I could get SIP clients for our phones, doesn't this mean using a data connection rather than just voice? That would make it a lot pricier than the current setup with DISA (which is largely free). /Per Jessen, Zürich A Wifi connection? I guess that wifi is not like it is here. I can get on highspeed wifi anywhere I go in the DC Metro area for free. In the cities, WiFi is typically only available in restaurants and cafes (Starbucks, McDonalds etc). In the country, no wifi. Well, the odd open access point, but using it is illegal, so that's a no-go. I would suspect that most road warriors have high speed data needs? Not sure what business you are in, but having fast internet (relatively speaking) is a must to do work. I am not saying to use the data supplied from phone, if that is what you are thinking. For my company, the mobile is primarily for voice - people don't spend that much time on the road, but when they do, they still want to appear as if they're in the office. If your phones don't have SIP, then use callback. You call your company, go through whatever you seutp in the dialplan, and the phone system calls you back as well as calling the other party. You edited out much of the context of the conversation to support your side. I don't play games like that... Sorry, that wasn't my intention, I just snip out the bits that aren't relevant to a reply. SIP client on the phone was an option. Was the original question about using DISA to save money? Yes it was. Now you are stating that it is largely free. I think the OPs question was about saving money, to which I suggested using DISA - it my setup it's largely free. Callback is a great solution when outbound cell phone calls quite a bit more than your cutrate VoIP provider. As I said, many countries do not charge for inbound calls. Right. I am still clueless what your point is/was but if it is almost free then, stick with it. Still clueless why you posted if it almost free. I did not post the original question, I just responded to it. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Linuxguy123 wrote: My original post didn't mention it, but I would like my home system to be Asterisk based. Has anyone figured out how to minimize cell charges when on the road via making calls via the home phone system ? Yep, look up DISA: http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA /Per Jessen, Zürich -- http://www.spamchek.com/ - fully managed email archive. Made in Switzerland. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Steve Totaro wrote: On Wed, Aug 24, 2011 at 2:42 AM, Per Jessen p...@computer.org wrote: Linuxguy123 wrote: My original post didn't mention it, but I would like my home system to be Asterisk based. Has anyone figured out how to minimize cell charges when on the road via making calls via the home phone system ? Yep, look up DISA: http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA /Per Jessen, Zürich -- Just curious how DISA would help with cell phone usage charges. Assuming multiple mobiles (e.g. household or office), a typical setup around here (Switzerland) is that you can call freely within a group of numbers, often including one or two fixnet numbers. But at least here, if you are on a per minute plan, how would DISA Where is here? help? Obviously, different countries and carriers do things differently, but I don't pay for anything extra, no roaming, nothing. Did you mean to say you don't pay for roaming either?? Wow. I could do with a subscription like that. (here roaming means using your phone in another country). For my situation, DISA is pointless except for road warriors who call all over the world, from anywhere, they can call into the corp system, get dialtone and skip the whole process of expense reports for work related calls. It makes things less complex, not more. Using DISA also means getting a corp caller id, not a mobile. Maybe if you explain your situation and how your plan works, but for me, personally, DISA would be a an added cost and complication. The only purpose I can think of for myself could be accomplished by spoofing caller id. How is that done from a mobile? Sofar that has been my main reason for using DISA - cost is not a real issue. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Per Jessen wrote: help? Obviously, different countries and carriers do things differently, but I don't pay for anything extra, no roaming, nothing. Did you mean to say you don't pay for roaming either?? Wow. I could do with a subscription like that. (here roaming means using your phone in another country). I guess theoretically roaming is using a GSM network other than your home network, but in Europe that = roaming internationally, which is typically very pricey. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS with Asterisk
Steve Totaro wrote: This link show how to send SMS using HTTP(s) and the format of the URL. http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN4201 The previous link is good news to me. Now I can do anything by hitting a URL. it is so simple. I've been asterisk's smsq for a couple of years: /usr/sbin/smsq --motx-channel='mISDN/2/062210' --motx-callerid=211 $1 $message /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] being bombarded with SIP packets
Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] being bombarded with SIP packets
Norbert Zawodsky wrote: Am 28.10.2010 09:41, schrieb Per Jessen: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich Hello Per, (iptables) rule #1: search the archives You will find nearly as many postings about that problem, as your server SIP packets received ... ;-) Thanks Norbert - I should take my own medicine, I'm usually the first to suggest searching the archives. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] being bombarded with SIP packets
Ishfaq Malik wrote: On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich Was it legitimate requests or a brute force attack? If it was a brute force attack have you considered using fail2ban? It appears to be brute force, but I haven't bothered to investigate any further. fail2ban is at best a kludge IMHO, and I don't like anything (automatically or otherwise) modifying my firewall. Like Nortbert suggested, I'll check the archives to see what others have done. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] being bombarded with SIP packets
Norbert Zawodsky wrote: Per, (didn't want to be unfriendly to you !) Not at all. As you say, you don't like anything to modify your firewal. My words! Someone (don't remember who when) on this list showed me a very clever trick (=iptables rule) to drop the packets if too many of them arrive within a given period of time. Works really great ! Yeah, I have a rule like that for SSH brute force attempts, and I did also find one for the same thing for SIP. Do not exatly remember how it was done (and I don't have access to that machine at the moment to have a look). I remeber something like first using iptables module string to inspect the packet if it contains the string REGISTER sip: and then use an iptables hash bucket with a limit of x/second This is what I found: iptables -N sip-flood iptables -A INPUT -p udp -m udp --dport 5060 -j sip-flood iptables -A INPUT -p tcp -m tcp --dport 5060:5061 --syn -j sip-flood iptables -A sip-flood -m recent --update --seconds 60 --hitcount 20 -j LOG --log-prefix SIP bruteforce attempt: iptables -A sip-flood -m recent --rcheck --seconds 60 --hitcount 20 -j DROP iptables -A sip-flood -m recent --set -j ACCEPT /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] being bombarded with SIP packets
Gordon Henderson wrote: On Thu, 28 Oct 2010, Norbert Zawodsky wrote: Am 28.10.2010 12:14, schrieb Per Jessen: Ishfaq Malik wrote: On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and This is not new - just Read The Fine Archives. Been going on for years. You're not the first, not the last. Well, to me it only started 3 days ago. Point taken though, I should have googled first. My main issue was not the brute force attempt in itself, but the increased latency it caused. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sms from Asterisk server
Johann Hoehn wrote: On 08/17/2010 09:00 AM, Tino wrote: Hello, I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful thanks This is easy to do by using email to SMS gateways. A list of them is on wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the Asterisk side, you have an extension that sends the email. I personally use an AGI script for this part, but you could use a System() call as well. Many telcos provide an SMSC, often also accessible over a landline. We use the Swisscom SMSC at 062210. (Swisscom subscription required). /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
Per Jessen wrote: Just start it with safe_asterisk. http://linux.die.net/man/8/safe_asterisk Unless my info is out of date, it will kill two birds with one stone. Asterisk will restart itself, and you will get a core dump. Thanks, Steve Totaro Hi Steve I've got three such core dumps now - do I just open a bugreport? See https://issues.asterisk.org/view.php?id=17178 /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson oej at edvina.net wrote: 7 feb 2010 kl. 15.09 skrev Per Jessen: Thomas Winter wrote: Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? My asterisk 1.4.23 also dies about once a month. I've never been able to work out why. I haven't seen this, but it is definitely something we should try to catch. It could be a memory leak or another type of leak. Any advice from other developers on how to try to catch this? One thing that would be good would be to get a core dump. There's a document in the /doc directory on how to recompile Asterisk with symbols and force a core dump to happen when we get a crash. /O Just start it with safe_asterisk. http://linux.die.net/man/8/safe_asterisk Unless my info is out of date, it will kill two birds with one stone. Asterisk will restart itself, and you will get a core dump. Thanks, Steve Totaro Hi Steve I've got three such core dumps now - do I just open a bugreport? /Per Jessen, Zürich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running safe_asterisk
Tilghman Lesher wrote: On Tuesday 23 February 2010 05:27:55 Per Jessen wrote: To be honest I don't remember any more, I just know my queueing doesn't work unless I reload. I think it's a timing issue at startup - that app_queue gets loaded too early or something. ah, here is my question about the same, but back in 2007: http://lists.digium.com/pipermail/asterisk-users/2007-May/188072.html You need to load the chan_local.so channel before pbx_config.so loads, so that your Local channels have the right devicestate. Adding 'preload = chan_local.so', followed by 'preload = pbx_config.so', to your /etc/asterisk/modules.conf should be sufficient. Thanks Tilghman - that works! I also added chan_sip.so. /Per Jessen, Zürich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Running safe_asterisk
About two weeks ago there was a thread about asterisk suddenly dying - I posted a response that the same happens to my asterisk about once a month, sometimes more. Someone suggested using 'safe_asterisk' (and get hold of a core dump) which sounds like a good idea, but one thing I can't figure is how to get module reload app_queue executed automatically at startup? /Per Jessen, Zürich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running safe_asterisk
Tzafrir Cohen wrote: On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote: About two weeks ago there was a thread about asterisk suddenly dying - I posted a response that the same happens to my asterisk about once a month, sometimes more. Someone suggested using 'safe_asterisk' (and get hold of a core dump) which sounds like a good idea, but one thing I can't figure is how to get module reload app_queue executed automatically at startup? All modules are loaded at startup. Why would you need a reload? To be honest I don't remember any more, I just know my queueing doesn't work unless I reload. I think it's a timing issue at startup - that app_queue gets loaded too early or something. ah, here is my question about the same, but back in 2007: http://lists.digium.com/pipermail/asterisk-users/2007-May/188072.html /Per Jessen, Zürich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC-S card
Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if that helps. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC-S card
Tzafrir Cohen wrote: On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote: On 2/22/2010 10:26 AM, Per Jessen wrote: Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if that helps. /Per Jessen, Zürich I have use this howto http://www.voip-info.org/wiki/view/Asterisk+zaphfc; , but i can´t put the card working in nt ptp mode. Can you explain me how i have to do that? Do you have any howto to make the card work in nt ptp mode? thanks for answer Short answert: signalling = bri_net Longer answer: That page is outdated (hmm, and I didn't get to update it :-( ) Nowadays (as of Asterisk 1.6.0) BRI support is included in Asterisk. The zaphfc driver, though, is still not included in DAHDI. It's maintained, though. The version included in the Debian packages is taken from http://git.tzafrir.org.il/?p=dahdi-extra.git;a=summary . Either way (bristuff or Asterisk = 1.6.0) to use BRI PTP NT in chan_dahdi you should set: signalling = bri_net for the span's channels in /etc/asterisk/chan_dahdi.conf . None of the above looks very familiar - I'm using Asterisk 1.4.x + misdn, one HFC-4S for the external lines and one plain (Conrad) HFS-PCI in NT mode for an ISDN DECT base-station. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
Thomas Winter wrote: Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? My asterisk 1.4.23 also dies about once a month. I've never been able to work out why. /Per Jessen, Zürich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Running a script after Dial() ?
I have the following dialplan:
; calls prefix by '8' are recorded
exten = _8[01]./_251,1,Set(something=shortened)
exten = _8[01]./_251,n,Set(WAV=filename)
exten = _8[01]./_251,n,Monitor(wav,${WAV},mb)
exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g)
exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav
${EXTEN:1} emailaddr)
exten = _8[01]./_251,n,Hangup()
The idea is that the caller may opt to record a conversation by
prefixing the dialled number with '8'. The wav file would then be
emailed to him when the call finishes.
The recording works fine, but the emailing doesn't - only when the
called party hangs up first, but if the caller hangs up, the
System(script) isn't called. What am I missing here?
/Per Jessen, Zürich
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Re: [asterisk-users] Running a script after Dial() ?
Steve Edwards wrote:
On Thu, 4 Feb 2010, Per Jessen wrote:
; calls prefix by '8' are recorded
exten = _8[01]./_251,1,Set(something=shortened)
exten = _8[01]./_251,n,Set(WAV=filename)
exten = _8[01]./_251,n,Monitor(wav,${WAV},mb)
exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g)
exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav
${EXTEN:1} emailaddr)
exten = _8[01]./_251,n,Hangup()
The idea is that the caller may opt to record a conversation by
prefixing the dialled number with '8'. The wav file would then be
emailed to him when the call finishes. The recording works fine, but
the emailing doesn't - only when the called party hangs up first, but
if the caller hangs up, the System(script) isn't called. What am I
missing here?
Check out the h exten.
Yes, I did look at that. The issue is that the above has a hardcoded
extension-to-emailaddr relation (251-emailaddr). I don't see how I can
use the h extension when the email-address will depend on the callers
extension. Well, I guess I could handle it in the script, but I'd
rather keep that information out of it.
/Per Jessen, Zürich
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Re: [asterisk-users] Running a script after Dial() ?
Danny Nicholas wrote:
Set the emailaddr into a channel variable. Since I'm there, just make
your h exten do the system if ${WAV} and ${emailaddr} are longer than
1. Like this.
- exten = h,1,noop(hangup logic)
- exten = h,n,Gotoif($[${LEN(${WAV})} 4]?just_hangup)
- exten = h,n,Gotoif($[${LEN(${emailaddr})} 4]?just_hangup)
- exten = h,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1}
${emailaddr})
- exten = h,n(just_hangup),Hangup
--
Danny Nicholas
Thanks Danny.
/Per Jessen, Zürich
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Re: [asterisk-users] Running a script after Dial() ?
Ben Dinnerville wrote: Why dont you use the MixMonitor application which allows for a system command to be passed in as an argument that is executed once the recording is finished??? - MixMonitor(file.ext[|options[|command]]) command will be executed when the recording is over. I did briefly consider that, but thought what I was trying to do should be perfectly feasible. Sorry, Monitor also has the flag param which allows you to execute a command post recording if you want to stick with Monitor and not MixMonitor. The monitor() works perfectly well, my only issue was getting a command executed after the recording was done. I might take another look at mixmonitor(), but as I've got it working now, and it's not exactly going to be used a lot, well ... /Per Jessen, Zürich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL syntax error : I really don't see where...
jonas kellens wrote: [Aug 30 14:07:42] -- Executing [...@macro-vakantie:2] MYSQL(IAX2/zoiper-9238, Query resultid 1 SELECT vakantie_set vakantie_data1 vakantie_data2 FROM AstDB where SIPACCOUNT=092779077) in new stack [Aug 30 14:07:42] WARNING[26029]: app_addon_sql_mysql.c:335 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'vakantie_data2 FROM AstDB where SIPACCOUNT=092779077' at line 1 You need to separate the selected columns with commas. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ -- 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] SPA3102 - How to save config in a file
Stefan Schmidt wrote: hello, you could retrieve the config from you SPA with the following url: http://ipofyourphone/admin/spacfg.xml . That works well with the Linksys phones, but not with the SPA-3102 which isn't really a phone, but an ATA. My 3102 has software version 5.1.6. /Per Jessen, Zürich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 - How to save config in a file
Jimmy Godbout wrote: Hi, The format of the file for the provisioning is xml. You create a file with the configuration you want and put it on your provisioning server. Then, you put a rule in the spa3102 to retrieve the file when the unit boot up. Well, with the other Linksys devices (SPA-941 etc), you can retrieve the configuration from the phone first - I think that's what the OP had in mind. /Per Jessen, Zürich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many bounces does it take before you get unsubscribed?
I got notified on 22 Jan that I was about to be unsubscribed due to excessive bounces. I've checked my mail logs, and saw the following bounces (that I had generated): Jan 12 04:50:44 437 Bad Message-ID Jan 16 19:56:19 437 Bad Message-ID Jan 19 20:36:20 437 Bad Message-ID Jan 22 18:11:35 437 Article posted in the future So over 10 days I had 4 bounces - I guess that's enough, but shouldn't the counter be reset at some point? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nightly tarballs, would you use them?
Russell Bryant wrote: Greetings, During the past week, there have been some requests for nightly tarballs to help making testing new Asterisk code easier. There was some debate as to whether they would be useful. The reason that they may not be useful is because you can get equivalent access to new code just by accessing the subversion repository directly. However, for one reason or another, some people would prefer to have a tarball. If this was available, would you be interested in it? On occasion, yes. I think nightly tarballs could be quite useful. Whilst it's easy to check out from subversion directly, a nightly tarball provides a specific point of reference which can be helpful when trying to identify a problem. If we had a specific problem we were trying to fix, I would very likely grab the latest tarball and try it out. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
Tilghman Lesher wrote: And finally, another person has already made the point that most XML editors are graphical in nature. A great many Asterisk installations are installed in locations where a graphical front end is not practical. ssh -X will deal with that. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
Ryan Burke wrote: I just was looking over the app_waitutil.c and am confused you add 500 to tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec + 500) / 1000);? Without having looked at Philips code at all, that looks like he is rounding up? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iptables requirements for SIP
Alejandro Cabrera Obed wrote: Does iptables need any SIP special module or something like this in order to let SIP+RTP work OK ??? No, you don't need anything special for iptables. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems getting Asterisk to detect call in SUSE9.3, with FritzCard
Frank Church wrote: I have installed an Asterisk 1.4 on Suse93 using a FritzCard. Some calls are logged to the ISDN log, but Asterisk is not detecting incoming calls. I wonder whether some other device or process is preventing Asterisk from gaining access to the isdn line? Frank, have you looked at using misdn? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gigaset S450ip and simultaneous calls
Olivier wrote: Hi, My Gigaset S450ip allows 2 simulatneous calls when each incoming call are targeted to different phones. When both calls target the same extension, the second one is forwarded to voicemail. So when the user is busy, you get voicemail - is that in your dialplan or is it the Gigaset that does it automagically? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building and running mISDN for B410P on Ubuntu 7.04
[EMAIL PROTECTED] wrote: Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Not being able to build mISDN on Ubuntu using make b410p I have used mISDN-1_1_7 which seems to work ok. QUESTION: Should I expect this version of mISDN to work ok with these cards? Or is there a way to build using make b410P on Ubuntu? (make force does not help at all) There is an misdn-asterisk mailing where you might have a better chance of getting a useful answer: Misdn-asterisk mailing list [EMAIL PROTECTED] http://lists.beronet.com/cgi-bin/mailman/listinfo/misdn-asterisk /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change the Voice promps in asterisk 1.4
voip crazy wrote: Hello all, Which is the best way to change the default Voice promps in asteriosk 1.4from english to french? You obtain/record new voice recordings, and add those under /var/lib/asterisk/sounds/fr/. Maybe someone has done this already and you can borrow their recordings? And if I would like to add a new Voice promp set, how is the way to do? I have used the recording system in voicemail, and then just moved the messages from /var/spool/asterisk/voicemail/mailbox to where I needed them. Not sure if this is the best way, but it does the job. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Traditional' Faxing
Jonn R Taylor wrote: There are alot of option for handeling faxes. One is to use iaxmodem and hylafax. This option works the best. Completely agree - we've been using such a setup for almost a year now. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] grandstream troubles
I've got a Grandstream 487 in a home-office. The phone-side is working fine, but the user is complaining that his internet connection keeps disappearing. The Grandstream is set up as NAT router, and there's just one PC hanging off the LAN. Has anyone experienced anything similar? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What do you do to keep asterisk alive?
I've asterisk stop (presumably segfaulting) a couple of times, and I was just beginning to look at how to keep it running - what have others done? I was thinking of wrapping a script around asterisk like this: while 1 do asterisk -f done /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you do to keep asterisk alive?
Tzafrir Cohen wrote: while true do asterisk -f done And if Asterisk decides to die? If you have a wrong module in /var/lib/asterisk/module ? Well, if asterisk decides to die, I want to restart it. A bad module would be spotted prior to going into production. You're reimplementing safe_asterisk badly. And safe_asterisk is bad enough as-is. So what do you do? I'd rather not have some elaborate monitoring and restart scheme. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you do to keep asterisk alive?
Andrea Spadaccini wrote: IMHO it's better to build a FSM (Finite State Machine) that handles the Asterisk process and other collateral processes (like the MAPI proxy) and let it monitor the process. Moreover, you should make this FSM sensible to UNIX signals in order to start, stop, restart Asterisk easily if you want. Hi Andrea that's exactly the kind of elaborate scheme I was hoping to avoid. What do you do today? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you do to keep asterisk alive?
Tony Mountifield wrote: Have a look at the safe_asterisk script, which should automatically be in /usr/sbin/safe_asterisk. It does this automatically, including emailing a notification (if you set the NOTIFY variable). Thanks, I didn't know that script (well, until Tzafrir mentioned it :-) /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you do to keep asterisk alive?
C F wrote: Why is it stooping on you? What version are you running? Are you running any AGI scripts? I don't know why it's stopping, but I'm pretty certain it's a segfault. Next time it happens, I should be getting the core dump. I'm running 1.4.13, no AGI scripts. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA-941 Unavailable
Kim Joung-il wrote: IP is changing because it is simply an public dynamic IP address, and our provider change the IP every 8 hours 1) is the phone set up as being behind a NAT router? 2) have you got a STUN server? I have a couple of SPA-921s in just such a setup with no problems. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] grandstream troubles
[EMAIL PROTECTED] wrote: Have you tried a second unit? I don't trust the Grandstream ATA at all. We only bought 3 but none worked! Nope, just the one. It's really a temp solution, so I don't want to stock up on them. Also, it has worked fine previously, albeit in a different location. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you do to keep asterisk alive?
Sajith T S wrote: It certainly isn't a replacement for fixing the root causes of whatever that makes asterisk die, though. Completely agree. I intend to have a look at that when it happens next time. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you do to keep asterisk alive?
Matthew J. Roth wrote: Per Jessen wrote: I don't know why it's stopping, but I'm pretty certain it's a segfault. Next time it happens, I should be getting the core dump. I'm running 1.4.13, no AGI scripts. Per, You should be able to determine if it was a segfault by looking at your system log. For example, on one of my CentOS systems: [EMAIL PROTECTED] ~]# grep segfault /var/log/messages Jul 31 11:10:41 ast01 kernel: asterisk[11548]: segfault at 0010 rip 00420450 rsp 41c670e0 error 4 Hi Matthew I've checked my logs, and there's nothing like that - but I don't think I've ever seen a logline being written because of a segfault. I wonder if it's a special config option. Hopefully, you don't have enough segfaults occurring on your system to require a more complex search. Nah, right now asterisk is the only service that is a little unstable. I also recommend the safe_asterisk script if your failures are all segfaults. Unfortunately, the other way Asterisk dies (the process is still alive, but not really doing anything) isn't as easy to diagnose. an strace might reveal what it's doing - or where. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Arbitrary limit on length of email address?
Alan Lord wrote: Whereas this one: [Nov 5 18:36:02] DEBUG[2519]: app_voicemail.c:1957 sendmail: Sent [mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' fails to get delivered and is 34 characters long. Both email accounts work otherwise and I have had no recorded problems with mails not arriving at the 34ch address before. Any ideas? Am I barking up the wrong tree? Check your mail-logs. Was the email with the long address accepted and processed by your mail-server? Also look for traces of an incomplete email-address being used (or something like that). /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML file for spa devices
Rizwan Hisham wrote: Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. For SPA-921 and SPA-941, you can get it from the phone itself: http://phoneip/admin/spacfg.xml I'm sure the same goes for SPA-961, but I don't have any of those. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
Jared Smith wrote: On Sun, 2007-10-21 at 13:42 +0200, Per Jessen wrote: The SPA-9x1 does support http download, but I don't see how you could change the initial TFTP request to HTTP without manually configuring the phone. Even then I'm not sure it would work - I certainly haven't managed to make any of my SPAs do an auto-config over HTTP. Actually, it's really easy to do. Here's a copy of my spa942.cfg file which I use to point the phone at my web server, as well as upgrade the firmware. Now that I've found my typo, I completely agree :-) I had $MAC instead of $MA, which produces a MAC address in the nn:nn:nn format. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT traversal packet loss measurement
Yitzhak Bar Geva wrote: How can one measure the effect of NAT traversal packet loss? We currently have no solution for NAT traversal for our SIP clients. We've recently completed a setup (see other threads) with a couple of SIP clients behind NAT in their respective home-offices. Took a couple of attempts, but after consulting the list, we have a working setup. What's the simplest method of preventing packet loss due to NAT traversal in a SIP environment? I doubt very much if any loss you're seeing is due to NAT traversal. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
Luki wrote: Here's how you do it. 1) In the DHCP server's config (dhcpd.conf) you specify the IP of the TFTP server: option tftp-server-name 66.55.44.33; This can be a remote server, as long as it's accessible by the device. 2) The factory settings on the Sipura devices (ATAs and phones) have /spa$PSN.cfg in the Provisioning profile rule, so the device will connect the TFTP server you specify and will try to retrieve that file, i.e. ftfp://66.55.44.33/spa942.cfg for the SPA-942 in this example. 3) This file contains very minimal information, which tells the device where to download its final configuration from. This can be a remote http server so you can maintain the configs on one central server. Example: flat-profile Profile_Rule ua=na http://YOUR.HTTP.PROVISIONING.SERVER.HOST/$MA.bin /Profile_Rule /flat-profile Is it possible that this only works with a compiled config? I've been trying do the above, but with the XML config, and I'm not getting anywhere. 4) The device will then connect via HTTP and will try to retrieve for configuration for its MAC address. Since it's a HTTP request, you can generate the provisioning data on the fly (even from the a database), either in XML format or in compiled format if you have the Sipura compiler. Oh well - I wonder what I'm doing wrong then. I've been trying to get this to work for most of last week. The above works just fine and very reliably. We have disabled periodic resync as the Sipura phones seem to reboot sometimes for no good reason when they apply the new but unchanged profile. If there is a config change, we just push it on the phone with SIP NOTIFY option. Do you push it from Asterisk or somewhereelse? Again, I can't make it work. I've got Auth resync-reboot disabled on the SPA, so it shouldn't be asking for authentication, but the SIP NOTIFY goes out, and the phone does nothing. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
Per Jessen wrote: Luki wrote: Here's how you do it. [snip] Oh well - I wonder what I'm doing wrong then. I've been trying to get this to work for most of last week. Luki, thanks for writing to say it DOES work. I've have just now had another look, found my mistakes (basically $MAC instead of $MA), and it's working! Do you push it from Asterisk or somewhereelse? Again, I can't make it work. I've got Auth resync-reboot disabled on the SPA, so it shouldn't be asking for authentication, but the SIP NOTIFY goes out, and the phone does nothing. Got that working too. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
[EMAIL PROTECTED] wrote: If you are trying to use non-complied (XML) profiles... don't even bother wasting your time. Oh. I _am_ using the XML format. When I initiate a resync over the http server, it works fine, except the SPA doesn't start the regular resync. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
Anselm Martin Hoffmeister wrote: Am Samstag, den 20.10.2007, 22:58 -0700 schrieb Philip Prindeville: I'd like to be able to templatize a server, add a bunch of new handsets into sip.conf and extensions.conf, and then plug the phones into a network and have some DHCP and/or TFTP glue logic that sees the DHCP or TFTP request, and from it generates a boot file (an .XML file) and a response parameter list for DHCP... populates a file into the /tftpboot/ directory, etc. How viable is this? The problem there is that you have a very small windows. AFAIK there are no tftp servers that can generate files on-the-fly, so your script would have to generate the XML within less than a second, reliably, and do all the necessary asterisk changes within another second or two, and I doubt this will be possible _that_ quick. Perhaps you could trigger the creation of the config, xml etc. on the first TFTP request - on the retry the files would then be ready to go. Of course you can use ISC dhcpd for tailoring answers to your needs (dynamic setting of config file etc), but IMO this will only work well if the phones support http config download, The SPA-9x1 does support http download, but I don't see how you could change the initial TFTP request to HTTP without manually configuring the phone. Even then I'm not sure it would work - I certainly haven't managed to make any of my SPAs do an auto-config over HTTP. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
[EMAIL PROTECTED] wrote: The SPA921 config has a NAT Keep Alive Intvl which is set to 15 by default, which I'm taking to mean it has NAT keep alives enabled. No, look under the Line 1 or Line 2 tab Found it - thanks again. Whilst I've got your attention - have you managed to make an SPA do a periodic config refresh? As far as I can tell, mine is all set to go, except it doesn't. I can poke it with the admin/resync URL, but I'd rather that it would ask on its own. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
Per Jessen wrote: Whilst I've got your attention - have you managed to make an SPA do a periodic config refresh? As far as I can tell, mine is all set to go, except it doesn't. I can poke it with the admin/resync URL, but I'd rather that it would ask on its own. I forgot to add - using http, not tftp. Every example I've seen talks about tftp, but I'm set up for http. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
Perssy Llamosas wrote: I doubt it. hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/ I think that is the sort of thing the OP would classify as religious grounds. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
Per Jessen wrote: [EMAIL PROTECTED] wrote: Did you set NAT Keep Alive Enable: = Yes for the line in question in the SPA's configuration? Uh, no, not specifically and I'm guessing it's not set by default? The SPA921 config has a NAT Keep Alive Intvl which is set to 15 by default, which I'm taking to mean it has NAT keep alives enabled. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
Russell Brown wrote: Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Linksys SPA94x/6x perhaps. I don't know if it has the transfer problem or not. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
[EMAIL PROTECTED] wrote: Did you set NAT Keep Alive Enable: = Yes for the line in question in the SPA's configuration? Uh, no, not specifically and I'm guessing it's not set by default? thanks. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Refrigerator Alarms
Balu Raman wrote: Omar, I am hoping that there may be some temp sensor interface that can be routed to a pc and if the temp falls out of a range, I can have this event call someone. I know what to do in asterisk to make a call. I have to do some research. may be, someone has already done a similar thing. Has to be event driven. Here's what we do - even it's not asterisk-related - temperatures are monitored/polled using Maxim/Dallas DS1820s devices. These are cheap and the size of a transistor. When/if certain thresholds are exceeded, an email is sent to our central mail-server where it is turned into an SMS. The same email could just as easily be turned into a call file and dropped into to the appropriate asterisk directory. We expect to start using SNMP traps instead of the email, but the principle is pretty much the same. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A linksys SPA921 behind NAT and firewall
I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 1-2. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side, coz' how would it know that UDP-traffic to SPA publicIP:5060 needs to be delivered to 192.168.x.x:5060 ? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. /Per Jessen, Zürich We use only openSUSE. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
Jay R. Ashworth wrote: On Thu, Oct 18, 2007 at 06:25:39PM +0200, Per Jessen wrote: Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. Well, no. /Per Jessen, Zürich We use only openSUSE. If A distro is a distro, then why? Because we like it, and because we're used to it. That we have an operational preference doesn't change a distro is a distro. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transferring callerid ?
I'm expanding our tiny asterisk setup with a couple of external SIP phones, and I've just noticed the issue of the callerid not being displayed on an attended transfer. This bug seems to deal with it: http://bugs.digium.com/print_bug_page.php?bug_id=8824 I'm surprised that this hasn't been dealt with a long time ago - is there perhaps a work-around that I'm not aware of? /Per Jessen, Zürich ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
Alan Lord wrote: I run a small Open Source consulting/training company here in the Uk and am starting to build an * server so that myself and my business partner (who both work from our respective homes) are communicating properly. I have a couple of colleagues who also work from home - they're hooked into our office telephone system (Asterisk box) using SIP phones from their respective home offices. This way they are virtually in the office - external calls can be forwarded 'internally', and when they call customers, it looks as if they're calling from the office. It also means that our main lines carry all the calling costs, so no extra bills or expenses to deal with. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callerids in UTF8 on SPA9x1? (was: Non-USASCII chars in sip.conf?)
Stefan Tichy wrote: On Fri, Sep 28, 2007 at 03:40:09PM +0200, Per Jessen wrote: This must have been asked before, but googling didn't help much. How do I define a callerid that contains non-USASCII characters? E.g. ä, ö, ü, å, ø, æ etc. ? Use UTF-8 Encoding. Thanks, why didn't I think of that ... Next question - does anyone know why such callerids in UTF8 would not display correctly on our Linksys/Sipura SPA9x1s ? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
Andrew Joakimsen wrote: That's horrible. I don't buy too many IP phones these days, but can anyone suggest a place better than the scumbags at VoIP supply? http://www.pcp.ch/ or http://www.digitec.ch/ /Per Jessen, Zürich ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Non-USASCII chars in sip.conf?
This must have been asked before, but googling didn't help much. How do I define a callerid that contains non-USASCII characters? E.g. ä, ö, ü, å, ø, æ etc. ? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Douglas Garstang wrote: Also be sure that you have a very redundant network configuration. Too often I see people spend a great deal of time and money to get redundant servers when their switches, firewalls, routers, etc are not even capable of handling a failed network element. You can achieve this at the application level. How do you do that when your single network connection is gone? When considering redundancy it is essential that you have no single point of failure. Depending on how far you want to go, this means right from your dual-box asterisk setup to dual diesel-generators and two multi-homed datacenters. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Atis Lezdins wrote: This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? Perhaps investigate/diagnose the craches? Software instability is not solved with a high-availability solution. IMHO. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Adrian Marsh wrote: I'm interested in how people are clustering Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I haven't looked into it in any detail, but how about the standard Linux HA solution with a heartbeat monitor, a shared file-system and IP take-over? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Philipp Kempgen wrote: Adrian Marsh wrote: so maybe it's a case of looking at Linux-HA. If I remember this correctly a normal ping is all Linux HA can do. It does not check whether Asterisk or other services are alive and respond to queries. I think the basic Linux-HA setup works with ping, but there's plenty of applications (mysql, apache, mailservers) that have their own plugins to monitor application level availability. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Philipp Kempgen wrote: I don't want to quote my text as not to spam the list (although it's all GPL). There's a nice countdown at http://www.amooma.de/gemeinschaft/ Very nice. I'll have to come back and take a closer look sometime. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use OpenVPN with Asterisk
MOSBAH ABDELKADER wrote: Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. 1. get openvpn up and running. That will give you a secure tunnel between server#1 and server#2. 2. whatever it is you need asterisk to do, make sure it uses the tunnel endpoints for networking. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searchable List Archives?
Matthew Rubenstein wrote: Maybe Digium could upgrade the list SW, or let me do it for them. Or I could set it up at my website, then import the list archive data and parse it into my DB for a searchable mirror. Assuming google is indexing the list archives at http://lists.digium.com/pipermail/asterisk-users you should be able search the archives using google. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searchable List Archives?
Per Jessen wrote: Matthew Rubenstein wrote: Maybe Digium could upgrade the list SW, or let me do it for them. Or I could set it up at my website, then import the list archive data and parse it into my DB for a searchable mirror. Assuming google is indexing the list archives at http://lists.digium.com/pipermail/asterisk-users you should be able search the archives using google. Search for SIP in the archives: http://www.google.ch/search?num=100hl=ensafe=offq=site%3Alists.digium.com+SIPbtnG=Searchmeta= /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RF to IP bridge
Curt Shaffer wrote: I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know there is an option available for the Avaya systems but its a little out of the price range Im looking for (~$200/channel). Has anyone out there found a stable way to do this? Radio-amateurs have done phone-patching for decades (where allowed) - there must be someone who can point you in the direction of an easy solution. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on OpenSuSE 10.2
Malcom Kemp wrote: Has anyone put Asterisk on the 10.2 distro? Any pointers? Yes, we're running 1.4.4 on openSUSE 10.2. We're have a couple of ISDN lines fed into each a TA card with a Cologne HFC chip. What you need to do is configure the before you jump to trying to build asterisk. This means running ./configure options in the asterisk source dir. It doesn't look like the README has been updated to include the configure step. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_exec: Unable to join queue
lenz wrote: I would try one of the two things: 1. adding a hint for the Local/[EMAIL PROTECTED] channels 2. using the = for queue members member = Agent/1001 member = Agent/1002 member = Agent/1003 Does this change anything? Hi Lenz thanks for your suggestions. I tried them both individually and together - no change. After a restart of Asterisk (now 1.4.4), I see the following on the first call with no callerid: app_queue.c:3541 queue_exec: Unable to join queue 'enidan' I then do a module reload app_queue, and everything is working fine. A show queue before reloading: enidan has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet No Callers Does anyone know what (Invalid) means in this context? I'll check the code myself, but just in case someone recognises it. After a reload: enidan has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet No Callers /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
Alex Balashov wrote: On Wed, 16 May 2007, Stephen Bosch said something to this effect: The fax-to-e-mail services charge as much as the telco does for a business line, sometimes more (at least, the ones I can deal with in this area). Better to set-up hylafax, IMHO. Not necessarily, except perhaps in cases of very high volumes. Actually, I think hylafax+iaxmodem are particularly useful for small volumes - the kind of situation where you do need a fax, but you might only receive or transmit something once a week at most. Setting up hylafax+iaxmodem takes about an hour. I haven't found any drawbacks yet. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Hardware Phone
Chris Bagnall wrote: We've had good results with the following (in ascending price order, as per prices in the UK): Elmeg IP290 (almost identical to the old Snom 190) Linksys SPA942 Aastra 57i Linksys SPA962 We're using only Linksys - 921s and 941s. When I was researching prices, I found I could buy them off a local Swiss retailer, and sell them on at ebay in the UK - for some reason the UK prices were significantly higher. This was about two months ago - today: SPA921 - SFr124 - roughly GBP51 (incl.VAT) SPA941 - SFr136 - roughly GBP56. (incl.VAT) On ebay.co.uk, the SPA-921 is selling for almost twice the price above. I just don't get it. I don't know the street-price though. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function_db_read: DB requires an argument, DB(family/key)
Lee Jenkins wrote:
OK, so I tried this:
exten = _X.,1,Noop(CallerId is ${CALLERID(all)})
exten = _X.,n,Noop(blurp)
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
This now appears to execute the first Noop(), skip the second, and
then issue the no argument warning on the Set() call.
Try an Answer() first?
OK, tried that, didn't change anything.
What I still don't get is - why does reloading the app_queue module fix
this problem? The app_queue issue is another one, but I just can't see
how it would influence the workings of the DB() function.
/Per Jessen, Zürich
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Re: [asterisk-users] Segmentation fault
Adam Lovegrove wrote:
Asterisk is crashing about once a day with segmentation fault.
This is the error..
/usr/sbin/safe_asterisk: line 111: 3482 Segmentation fault (core
dumped) nice –n $PRIORITY ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
Is this information helpful?
Can anyone suggest anything?
Can I provide anymore useful information for troubleshooting?
I think it would be useful if you could describe what the system is
doing when it crashes - if you know. Also, the core dump will probably
help someone diagnose the problem. (but don't send it to the list :-)
/Per Jessen, Zürich
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Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
Olivier wrote: Do you mean nobody has ever done this before (as I thought before asking this question to the list) ? So which tool KDE users are using for this ? I use vi(m). /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_exec: Unable to join queue
lenz wrote: Is the queue enidan configured at all in queues.conf? and how is it defined? l. Sorry, I should have added that: from queues.conf: [enidan] strategy = ringall ;announce = enidan-queue member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] Also, what I discovered yesterday is the following: just after an asterisk restart: *CLI show queue enidan has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet No Callers The (Invalid) bit is worrying, but after a reload of app_queue: *CLI show queue enidan has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet No Callers /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function_db_read: DB requires an argument, DB(family/key)
Per Jessen wrote:
from extensions.conf:
exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I basically try to lookup the CLIP and attach a name for each inbound
call. This works fine, except when I have just restarted asterisk -
at which time I've more than once seen the message from the subject.
As far as I can tell, with my Set(CALLERID), I should always have an
argument in the DB function?
Is there a better/more appropriate place/list to ask this kind of
question?
/Per Jessen, Zürich
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Re: [asterisk-users] function_db_read: DB requires an argument, DB(family/key)
Gordon Henderson wrote:
You're getting the error message because ${CALLERID(num)} is empty.
ie. there is no caller-Id set, so I'd work on working out why there's
no callerId set for the very first call...
Eg. start with:
exten = _X.,1,Noop(CallerId is ${CallerId(all)})
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I do omething similar, but I test for no callerId before trying to do
a database lookup.
Later on I have no problems with e.g. a suppressed callerid - but I'll
try what you suggest. Thanks Gordon.
OK, tried it -
with your Noop(), I don't get a warning when there is no CLIP:
-- Executing [EMAIL PROTECTED]:1] NoOp(mISDN/3-u0, CallerId is
) in new stack
-- Executing [EMAIL PROTECTED]:2] Ringing(mISDN/3-u0, ) in new stack
What surprises is that my Set() call isn't listed in the console log?
OK, so I tried this:
exten = _X.,1,Noop(CallerId is ${CALLERID(all)})
exten = _X.,n,Noop(blurp)
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
This now appears to execute the first Noop(), skip the second, and then
issue the no argument warning on the Set() call.
And miraculously, I can make the whole thing work by issuing a
module reload app_queue. After that, the DB() function no longer
complains, with or without CLIP.
Sounds like a bug to me.
/Per Jessen, Zürich
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Re: [asterisk-users] Dry Copper Pair
Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: C F wrote: Stephen i disagree. growing up in new work city i can say its quite easy to get away with it in the city. where i live now in new jersey (population of around 6) i wouldnt be able to pull that off. The world is a big place, and I suppose there's room for all kinds. In these parts, the vigilance is pretty high. The pillars are padlocked now; they didn't use to be, and the COs are locked down like Fort Knox. Anyway, I know enough more than one person who has landed in the clink for treating the telco like a personal lab. what exactly was the charge ? Perhaps something along the lines of unauthorised tampering with a telecomms installation? /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] function_db_read: DB requires an argument, DB(family/key)
from extensions.conf:
exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I basically try to lookup the CLIP and attach a name for each inbound
call. This works fine, except when I have just restarted asterisk - at
which time I've more than once seen the message from the subject.
As far as I can tell, with my Set(CALLERID), I should always have an
argument in the DB function?
/Per Jessen, Zürich
--
ENIDAN Technologies GmbH - managed email security.
Starting at SFr1/month/user - http://www.spamchek.ch/
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[asterisk-users] queue_exec: Unable to join queue
I have a queue defined which I use like this: exten = _X.(reception),n,Ringing exten = _X.,n,Queue(enidan,t,,,20) exten = _X.,n,Voicemail(443,u) exten = _X.,n,Hangup() When I start asterisk, this queue doesn't work - -- Executing [EMAIL PROTECTED]:3] Queue(mISDN/3-u0, enidan|t|||20) in new stack [May 14 13:53:59] WARNING[17860]: app_queue.c:3541 queue_exec: Unable to join queue 'enidan' -- Executing [EMAIL PROTECTED]:4] VoiceMail(mISDN/3-u0, 443|u) in new stack But all I need to do to fix it is reload app_queue. Does anyone know what's going on? /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
Stephen Bosch wrote: # cat /proc/interrupts CPU0 CPU1 1: 1626 0Phys-irq i8042 6: 3 0Phys-irq floppy 8: 0 0Phys-irq rtc 9: 0 0Phys-irq acpi 14: 63 0Phys-irq ide0 16: 1 0Phys-irq libata, eth3 17:6762583 0Phys-irq libata 18: 13789 0Phys-irq libata 19: 33459690 0Phys-irq eth1 20: 19864325 0Phys-irq sky2, eth0 21: 269250881 0Phys-irq wctdm 256: 77735119 0 Dynamic-irq timer0 257:3986325 0 Dynamic-irq resched0 258: 37 0 Dynamic-irq callfunc0 259: 04652748 Dynamic-irq resched1 260: 0139 Dynamic-irq callfunc1 261: 0 28924306 Dynamic-irq timer1 262: 1021 0 Dynamic-irq xenbus I've never seen cat /proc/interrupts output that looks like that... waaaitaminute... are you running this in a virtual machine? Or on a machine running virtual machines? It looks like a XEN machine. Well spotted, Stephen. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
Atlanticnynex wrote: whether Asterisk could handle roughly one DS3's worth of calls (672 calls) just doing the LCR (I've seen some pre-built LCR apps, looks like they all do on-the-fly MySQL queries- I think I'd write my own AGI that would use a cache). When appropriately configured, MySQL does a pretty good job of caching results too. [129 lines snipped] /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Sven Jacobs wrote: Dear users, I think I may found a bug in the voicemail module of Asterisk 1.4.2! Outgoing email notifications should use a real existing domain (let's call it domain.real) instead of the local domain (domain.local) so that some mail servers won't reject the mails. That's why I've set the serveremail option in voicemail.conf to [EMAIL PROTECTED] Unfortunately Asterisk is always sending these mails with the sender [EMAIL PROTECTED] regardless of the serveremail option. You fix that in your mail-server with aliasing and/or canonicalising. I think the Asterisk behaviour is correct. It is similar to receiving an email from cron or some other daemon. That is sent from [EMAIL PROTECTED], which is fine for your internal purposes, but if you send it out externally, you'll need to map it to a external address. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Sven Jacobs wrote: You fix that in your mail-server with aliasing and/or canonicalising. I think the Asterisk behaviour is correct. It is similar to receiving an email from cron or some other daemon. That is sent from [EMAIL PROTECTED], which is fine for your internal purposes, but if you send it out externally, you'll need to map it to a external address. But then again I don't understand the serveremail option. What is it for then? As far as I can tell (but I'm on 1.4.1), the serveremail option only sets the From-address, not the envelope-address. The envelope will probably always be asterisk-user@hostname /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Sven Jacobs wrote: As far as I can tell (but I'm on 1.4.1), the serveremail option only sets the From-address, not the envelope-address. The envelope will probably always be asterisk-user@hostname The From-address ist set by the fromstring option - which works btw - so you are wrong :) Unfortunately setting the From-address does not fix my problem. Maybe I'm misinterpreting things, but this is what I se: fromstring = the From:-text, not the From:-address. I'm just using the default fromstring, but I've set serveremail = asterisk@realdomain With this I get From: Asterisk PBX [EMAIL PROTECTED] Still, the envelope is [EMAIL PROTECTED]. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Joshua Colp wrote: The voicemail email gets handed off to sendmail for actual sending. It's adding on the envelope above. Yes, but asterisk is writing the From: header. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
