Re: [asterisk-users] Losing local SIP phones when internet goes down?
On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson o...@edvina.net wrote: What I have seen on my asterisk box when I had a up/down adsl line was that the asterisk box couldn't do dns resolution and would hang( well no other internal calls could be made, seemed like some sort of semaphore was stuck) when the adsl came up and dns could be done, everything worked fine again Confirmed and experienced years ago in a release far, far away. Yes, that is the case. Asterisk doesn't have asynchronus DNS support, so in order to work when the link is down, you need a local resolver, like a caching BIND server, on the same host. The calls to DNS resolvers in Asterisk is synchronus, so Asterisk will wait for the response to arrive. IIRC, at the time I had this problem, asterisk did not answer analog phone lines either so as a company we had no phones and had to revert to regular telephones plugged into the wall. Even if Internet is working, if the configured DNS is down, you're still sunk. This sorely needs to be fixed IMO. /r -- _ -- 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] Losing local SIP phones when internet goes down?
On 05/02/10 09:15, Randy R wrote: On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson o...@edvina.net wrote: What I have seen on my asterisk box when I had a up/down adsl line was that the asterisk box couldn't do dns resolution and would hang( well no other internal calls could be made, seemed like some sort of semaphore was stuck) when the adsl came up and dns could be done, everything worked fine again Confirmed and experienced years ago in a release far, far away. Yes, that is the case. Asterisk doesn't have asynchronus DNS support, so in order to work when the link is down, you need a local resolver, like a caching BIND server, on the same host. The calls to DNS resolvers in Asterisk is synchronus, so Asterisk will wait for the response to arrive. IIRC, at the time I had this problem, asterisk did not answer analog phone lines either so as a company we had no phones and had to revert to regular telephones plugged into the wall. Even if Internet is working, if the configured DNS is down, you're still sunk. This sorely needs to be fixed IMO. /r Also had this many times, the installation and config of dnsmasq fixed it for my systems. DC smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Ongoing calls interface
I'm currently working on a PHP web interface to show (1) the registered endpoints and (2) their status: available, outgoing call or incoming call. (In the future, this interface should also be able to redirect calls etc.) (1) The interface already shows a list of all registered endpoints. For this, I had to work around Asterisk's delay in monitoring when an endpoint is unregistered. What I the interface does, is ping each endpoint to make sure it is still registered. Is there a better way to retrieve the endpoints still registered, instead of those that were registered an hour or so ago but might be unregistered already? (2) For each registered endpoint, the status needs to be displayed in the web interface. As I see it, there are two ways to retrieve this information. (a) Either I pull the information from Asterisk, by using some kind of core show channels command. I tried to use CDR to write calls made by the endpoints to a database. This works, but the rows are only written when a call is disconnected; I want to be able to see the status before that. (b) Another way is to keep track of notifications sent by endpoints. This is what I'm doing now. I added action URIs to all phones to make them send their data to a URL when they initiate a call, end a call etc. At that URL, a PHP script reads the data and saves it to a database. This works, but it feels like writing functionality that might already be available in Asterisk itself. Are there any other (better ways)? Which of the methods would be fastest for a large number of endpoints, and most reliable? Any help is greatly appreciated. With kind regards, David de Boer -- _ -- 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] Losing local SIP phones when internet goes down?
- Randy R randulo2...@gmail.com wrote: On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson o...@edvina.net wrote: What I have seen on my asterisk box when I had a up/down adsl line was that the asterisk box couldn't do dns resolution and would hang( well no other internal calls could be made, seemed like some sort of semaphore was stuck) when the adsl came up and dns could be done, everything worked fine again Confirmed and experienced years ago in a release far, far away. Yes, that is the case. Asterisk doesn't have asynchronus DNS support, so in order to work when the link is down, you need a local resolver, like a caching BIND server, on the same host. The calls to DNS resolvers in Asterisk is synchronus, so Asterisk will wait for the response to arrive. IIRC, at the time I had this problem, asterisk did not answer analog phone lines either so as a company we had no phones and had to revert to regular telephones plugged into the wall. Even if Internet is working, if the configured DNS is down, you're still sunk. This sorely needs to be fixed IMO. /r Why not run a internal DNS with forwarders to your ISP ? That way Asterisk can still resolve itself and hosts internally. -- Thanks, Phil -- _ -- 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] Losing local SIP phones when internet goes down?
5 feb 2010 kl. 09.28 skrev --[ UxBoD ]--: - Randy R randulo2...@gmail.com wrote: On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson o...@edvina.net wrote: What I have seen on my asterisk box when I had a up/down adsl line was that the asterisk box couldn't do dns resolution and would hang( well no other internal calls could be made, seemed like some sort of semaphore was stuck) when the adsl came up and dns could be done, everything worked fine again Confirmed and experienced years ago in a release far, far away. Yes, that is the case. Asterisk doesn't have asynchronus DNS support, so in order to work when the link is down, you need a local resolver, like a caching BIND server, on the same host. The calls to DNS resolvers in Asterisk is synchronus, so Asterisk will wait for the response to arrive. IIRC, at the time I had this problem, asterisk did not answer analog phone lines either so as a company we had no phones and had to revert to regular telephones plugged into the wall. Even if Internet is working, if the configured DNS is down, you're still sunk. This sorely needs to be fixed IMO. /r Why not run a internal DNS with forwarders to your ISP ? That way Asterisk can still resolve itself and hosts internally. See above: you need a local resolver, like a caching BIND server, on the same host. /O :-) -- _ -- 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] Losing local SIP phones when internet goes down?
Why not run a internal DNS with forwarders to your ISP ? That way Asterisk can still resolve itself and hosts internally. See above: you need a local resolver, like a caching BIND server, on the same host. Nice, but still, it ruins the all in one concept. Isn't there a lighter solution? /r -- _ -- 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] OpenVPN on phones?
Hi! OpenVPN by default uses UDP, but can be configured to use TCP. So what's the configuration on the Snom? Can I change it? Google is your friend: http://wiki.snom.com/Networking/VPN Philipp -- _ -- 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] Losing local SIP phones when internet goes down?
- Olle E. Johansson o...@edvina.net wrote: 5 feb 2010 kl. 09.28 skrev --[ UxBoD ]--: - Randy R randulo2...@gmail.com wrote: On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson o...@edvina.net wrote: What I have seen on my asterisk box when I had a up/down adsl line was that the asterisk box couldn't do dns resolution and would hang( well no other internal calls could be made, seemed like some sort of semaphore was stuck) when the adsl came up and dns could be done, everything worked fine again Confirmed and experienced years ago in a release far, far away. Yes, that is the case. Asterisk doesn't have asynchronus DNS support, so in order to work when the link is down, you need a local resolver, like a caching BIND server, on the same host. The calls to DNS resolvers in Asterisk is synchronus, so Asterisk will wait for the response to arrive. IIRC, at the time I had this problem, asterisk did not answer analog phone lines either so as a company we had no phones and had to revert to regular telephones plugged into the wall. Even if Internet is working, if the configured DNS is down, you're still sunk. This sorely needs to be fixed IMO. /r Why not run a internal DNS with forwarders to your ISP ? That way Asterisk can still resolve itself and hosts internally. See above: you need a local resolver, like a caching BIND server, on the same host. /O :-) Doh! :) My philosophy has always been to install a local named server, whether it be for Asterisk or something else, as most of the time everything I do is behind NAT and I prefer to resolve internal addresses. This also help if you run your own mailserver and make extensive queries to RBLs etc. -- Thanks, Phil -- _ -- 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] Losing local SIP phones when internet goes down?
- Randy R randulo2...@gmail.com wrote: Nice, but still, it ruins the all in one concept. Isn't there a lighter solution? Nice and lite DNS server ? http://www.nlnetlabs.nl/projects/nsd/ -- Thanks, Phil -- _ -- 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] Losing local SIP phones when internet goes down?
5 feb 2010 kl. 10.37 skrev Randy R: Why not run a internal DNS with forwarders to your ISP ? That way Asterisk can still resolve itself and hosts internally. See above: you need a local resolver, like a caching BIND server, on the same host. Nice, but still, it ruins the all in one concept. Isn't there a lighter solution? Why does it ruin all in one if it's not all without a local resolver? The only difference is that your resolver ask your ISP's DNS server for answers instead of Asterisk, since your resolver can handle the situation of not getting DNS replies properly. With caching, it might also speed up issues. You still need a mail server for running Asterisk, so adding a simple DNS server doesn't add much to the all in one list. /O -- _ -- 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] OpenVPN on phones?
5 feb 2010 kl. 10.36 skrev Philipp von Klitzing: Hi! OpenVPN by default uses UDP, but can be configured to use TCP. So what's the configuration on the Snom? Can I change it? Google is your friend: http://wiki.snom.com/Networking/VPN So what you're saying is that you have full access to the VPN configuration. That's cool. And typical of SNOM. My personal summary is that this is a good feature in a phone, if you have access to the detailed config. Thanks everyone for the feedback! /O -- _ -- 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] Losing local SIP phones when internet goes down?
On Fri, Feb 5, 2010 at 10:39 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: Doh! :) My philosophy has always been to install a local named server, whether it be for Asterisk or something else, as most of the time everything I do is behind NAT and I prefer to resolve internal addresses. This also help if you run your own mailserver and make extensive queries to RBLs etc. That last bit makes a good point. And speaking of RBL, is anyone doing a SPIT RBL? I was plagued by comment spam on an old forum I wrote in C years ago and I finally wrote a function to check projecthoneypot.org's httpBL. I feel like my days just gained an hour, the one I wasted every day modertaing useless spam comments. Are there lists to check for know pests in VoIP? /r -- _ -- 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] Ongoing calls interface
On Fri, Feb 5, 2010 at 9:38 AM, David de Boer boerdeda...@gmail.com wrote: I'm currently working on a PHP web interface to show (1) the registered endpoints and (2) their status: available, outgoing call or incoming call. (In the future, this interface should also be able to redirect calls etc.) (1) The interface already shows a list of all registered endpoints. For this, I had to work around Asterisk's delay in monitoring when an endpoint is unregistered. What I the interface does, is ping each endpoint to make sure it is still registered. Is there a better way to retrieve the endpoints still registered, instead of those that were registered an hour or so ago but might be unregistered already? (2) For each registered endpoint, the status needs to be displayed in the web interface. As I see it, there are two ways to retrieve this information. (a) Either I pull the information from Asterisk, by using some kind of core show channels command. I tried to use CDR to write calls made by the endpoints to a database. This works, but the rows are only written when a call is disconnected; I want to be able to see the status before that. (b) Another way is to keep track of notifications sent by endpoints. This is what I'm doing now. I added action URIs to all phones to make them send their data to a URL when they initiate a call, end a call etc. At that URL, a PHP script reads the data and saves it to a database. This works, but it feels like writing functionality that might already be available in Asterisk itself. Are there any other (better ways)? Which of the methods would be fastest for a large number of endpoints, and most reliable? Any help is greatly appreciated. You should do a google for Asterisk Manager Interface. :) -- _ -- 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] Losing local SIP phones when internet goes down?
Hi all, Many thanks for all your very fast and really helpful replies! Now I know about the asynchronous DNS issue, this all starts to make sense: presumably, when I disabled eth1 completely, the DNS queries just failed immediately, so didn't hold anything else up, whereas in the other scenarios (router or phone line down) the DNS requests weren't being answered, so Asterisk was waiting, preventing it from doing other things properly. So, as suggested, I'll go ahead and install dnsmasq (or similar), and that should fix things. Cheers, Nikhil. -- _ -- 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] Still on spandsp/app_fax and T.38
This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :) Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it might be spandsp-related. What happens is, when I use app_fax to receive a fax using T.38, I get very low transmission rates, and about 30% of the faxes are just not transmitted at all. 2400bps is not uncommon. If instead of T.38 I use an analog landline connected to that very same Asterisk box, app_fax works wonderfully well transmitting at full 9600bps. Just to rule out Asterisk's T.38 handling, I noticed that Fax For Asterisk works perfectly with T.38, also on that same box. I'll be happy to provide you any info that could help solve this issue. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 -- _ -- 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] Beginners Guide to setting up a Call Centre
On 16 January 2010 06:04, Sean Brady sbr...@gtfservices.com wrote: Looking at all the docs I can find Asterisks looks like it should be able to do the job and a whole lot more. This is for a small call centre so ideally we want all the features of an average call centre, ACD, Call Recording, Queue's etc etc. Any pointers on how to get started would be most helpful. Peter. --- (sorry this is so long) Peter, I figured that I would chime in, as I run IT and am a managing partner of a small call center based on Asterisk and I think that my experience will be helpful (hate to beat a dead horse)... Asterisk can definitely do what you need, so I am not going to talk about that any further. I wouldn't waste my time with anything else. I would strongly recommend either of the two following methods to get started, with the deciding factors being time and money. There are lots of factors that will sway this argument, such as the complexity of your workflow, CTI needs, etc., but those time and money are the biggies. You also have to carefully weigh your support requirements, uptime, and your desire to manage a phone system. Asterisk doesn't have to take that much work once it's installed and tuned, but it will require some maintenance. You will need to evaluate whether or not you want to take on that maintenance role or whether you want to pay to have it done for you. Method 1: A professional installation by a Digium Certified Asterisk Professional. It will cost you some money, how much depends on your needs and how clearly you articulate them. There are lots of great people out there that can help you get EXACTLY what you want and design a system that will grow with your business. Call Digium for recommendations, or reply to this with your contact info and we can talk off list (I'm not trying to sell anything, but I have some people that I can recommend). This can be a great option for a solid Asterisk system with good support and reliable operation with little maintenance. There's a couple different approaches to this method- managed and developed with support. Managed is where the team that developed the dialplan and asterisk environment for you manages the system for you as well for a recurring support fee. Drawbacks to this method: A. You will have to find a good vendor that will charge fairly and deliver on their SLA (always get an SLA with enforceable penalties). This isn't that tough, but it's important. B. The recurring support costs can eat into your budget quickly C. This will take some time to develop properly, and for simple environments it may be overkill. D. Adds/changes/ and deletes can be costly as well. This can be mitigating by communicating the need to accommodate staff turnover with a user maintainable system. Does not sound much worse than what we have now :) Method 2: Get a distro, install it, be dialing in about 8 hours or less (the route that I took when we started). This method is by far and away the easiest, cheapest, get-it-up-and-running-consequences-be-damned method. You will take less time, effort and money to get going like this than any other way I know of. If your call flow is simple to moderately complex, this is the way to go in my opinion. The FreePBX distros (Trixbox, AsteriskNOW [I think], Elastix, etc) all are very well put together, and will do everything that you listed in your original message and then some. Of the distro's, I would probably either go with AsteriskNOW or, if you are up for a little more setup work, FreePBX on it's own. Drawbacks to this method: A. I can't speak for others, but I found that the configuration engines have their limitations when it comes to call centers. They simply weren't designed to do some of the specific things that we needed to do as we grew. This doesn't mean that they wont do everything you need though, each case is unique. They were fine for us in the beginning, but as our business grew so did our specific needs, and we outgrew these solutions. There is nothing wrong with that if you understand from the outset that you may have needs that aren't met in the future. These distros have to factor in the needs of their respective communities, and what may be good for one organization might not be good for others. B. Troubleshooting issues can be more complex as you start to understand Asterisk and increase your level of sophistication. I had a hard time troubleshooting FreePBX until I understood it's dialplan more, and it made troubleshooting complicated as I didn't fully understand the call flow through it's dialplan. The more you work with it, the easier it gets, but there can be a learning curve. C. Integration with other vendor's products can sometimes be a challenge if they don't already support your
Re: [asterisk-users] Losing local SIP phones when internet goes down?
Hi again, OK, I've now installed a local caching nameserver, but don't see any change at all. IN detail, what I did: - Installed Debian packages resolvconf and dnsmasq (resolvconf just takes care of dynamic nameserver allocations in /etc/resolv.conf). - After looking at the docs, edited /etc/network/interfaces to add a dns-nameservers line in the entry for eth1. Then reconfigured resolvconf. - Checked /etc/resolv.conf: now showing 127.0.0.1 as the only nameserver. - Tested name resolution in general: working fine. - Turned ADSL router off and tried to make local and Zap calls: no luck. - Rebooted machine and tried again: still no luck. Again, the logs indicate that Asterisk thinks the SIP phones are unreachable. Was there anything special I needed to do with the setup of dnsmasq, or its interface with Asterisk? If not, I'm stuck again. Thoughts? Nikhil. On Fri, 5 Feb 2010, Nikhil Nair wrote: Hi all, Many thanks for all your very fast and really helpful replies! Now I know about the asynchronous DNS issue, this all starts to make sense: presumably, when I disabled eth1 completely, the DNS queries just failed immediately, so didn't hold anything else up, whereas in the other scenarios (router or phone line down) the DNS requests weren't being answered, so Asterisk was waiting, preventing it from doing other things properly. So, as suggested, I'll go ahead and install dnsmasq (or similar), and that should fix things. Cheers, Nikhil. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
On Fri, 5 Feb 2010, Nikhil Nair wrote: Hi again, OK, I've now installed a local caching nameserver, but don't see any change at all. IN detail, what I did: - Installed Debian packages resolvconf and dnsmasq (resolvconf just takes care of dynamic nameserver allocations in /etc/resolv.conf). - After looking at the docs, edited /etc/network/interfaces to add a dns-nameservers line in the entry for eth1. Then reconfigured resolvconf. - Checked /etc/resolv.conf: now showing 127.0.0.1 as the only nameserver. - Tested name resolution in general: working fine. - Turned ADSL router off and tried to make local and Zap calls: no luck. - Rebooted machine and tried again: still no luck. Again, the logs indicate that Asterisk thinks the SIP phones are unreachable. Was there anything special I needed to do with the setup of dnsmasq, or its interface with Asterisk? If not, I'm stuck again. Thoughts? Nikhil. Hi, I am stepping out on a limb here, since I have never run dnsmasq, but I don't think it is an actual caching server. I think it just relays queries to upstream servers, which in your case are still unreachable, and will still cause asterisk to timeout waiting for a reply. You need a true local DNS server that can answer for your asterisk box and any named phones. A caching server should do also, assuming that your link is up long enough to serve and cache a few local queries before it goes down - pretty much how most of my systems run. j -- _ -- 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] Quad Card PRI, Disable Unused Ports or Manage Channels. How?
[...@macro-record-enable:1] GotoIf(SIP/105-0f643cf0, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/105-0f643cf0, recordingcheck|20100205-132202|1265376122.2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20100205-132202|1265376122.2: Outbound recording enabled. recordingcheck|20100205-132202|1265376122.2: CALLFILENAME=OUT105-20100205-132202-1265376122.2 -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:999] MixMonitor(SIP/105-0f643cf0, OUT105-20100205-132202-1265376122.2.wav||) in new stack == Begin MixMonitor Recording SIP/105-0f643cf0 -- Executing [9...@from-internal:4] Macro(SIP/105-0f643cf0, dialout-trunk|1|150||) in new stack -- Executing [...@macro-dialout-trunk:1] Set(SIP/105-0f643cf0, DIAL_TRUNK=1) in new stack -- Executing [...@macro-dialout-trunk:2] GosubIf(SIP/105-0f643cf0, 0?sub-pincheck|s|1) in new stack -- Executing [...@macro-dialout-trunk:3] GotoIf(SIP/105-0f643cf0, 0?disabletrunk|1) in new stack -- Executing [...@macro-dialout-trunk:4] Set(SIP/105-0f643cf0, DIAL_NUMBER=150) in new stack -- Executing [...@macro-dialout-trunk:5] Set(SIP/105-0f643cf0, DIAL_TRUNK_OPTIONS=tr) in new stack -- Executing [...@macro-dialout-trunk:6] Set(SIP/105-0f643cf0, OUTBOUND_GROUP=OUT_1) in new stack -- Executing [...@macro-dialout-trunk:7] GotoIf(SIP/105-0f643cf0, 1?nomax) in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [...@macro-dialout-trunk:9] GotoIf(SIP/105-0f643cf0, 0?skipoutcid) in new stack -- Executing [...@macro-dialout-trunk:10] Set(SIP/105-0f643cf0, DIAL_TRUNK_OPTIONS=) in new stack -- Executing [...@macro-dialout-trunk:11] Macro(SIP/105-0f643cf0, outbound-callerid|1) in new stack -- Executing [...@macro-outbound-callerid:1] ExecIf(SIP/105-0f643cf0, 0|SetCallerPres|) in new stack -- Executing [...@macro-outbound-callerid:2] ExecIf(SIP/105-0f643cf0, 0|Set|REALCALLERIDNUM=105) in new stack -- Executing [...@macro-outbound-callerid:3] GotoIf(SIP/105-0f643cf0, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [...@macro-outbound-callerid:6] Set(SIP/105-0f643cf0, USEROUTCID=) in new stack -- Executing [...@macro-outbound-callerid:7] Set(SIP/105-0f643cf0, EMERGENCYCID=) in new stack -- Executing [...@macro-outbound-callerid:8] Set(SIP/105-0f643cf0, TRUNKOUTCID=) in new stack -- Executing [...@macro-outbound-callerid:9] GotoIf(SIP/105-0f643cf0, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [...@macro-outbound-callerid:12] ExecIf(SIP/105-0f643cf0, 0|Set|CALLERID(all)=) in new stack -- Executing [...@macro-outbound-callerid:13] ExecIf(SIP/105-0f643cf0, 0|Set|CALLERID(all)=) in new stack -- Executing [...@macro-outbound-callerid:14] ExecIf(SIP/105-0f643cf0, 0|SetCallerPres|prohib_passed_screen) in new stack -- Executing [...@macro-dialout-trunk:12] ExecIf(SIP/105-0f643cf0, 0|AGI|fixlocalprefix) in new stack -- Executing [...@macro-dialout-trunk:13] Set(SIP/105-0f643cf0, OUTNUM=150) in new stack -- Executing [...@macro-dialout-trunk:14] Set(SIP/105-0f643cf0, custom=DAHDI/g1) in new stack -- Executing [...@macro-dialout-trunk:15] ExecIf(SIP/105-0f643cf0, 0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)) in new stack -- Executing [...@macro-dialout-trunk:16] Macro(SIP/105-0f643cf0, dialout-trunk-predial-hook|) in new stack -- Executing [...@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/105-0f643cf0, ) in new stack -- Executing [...@macro-dialout-trunk:17] GotoIf(SIP/105-0f643cf0, 0?bypass|1) in new stack -- Executing [...@macro-dialout-trunk:18] GotoIf(SIP/105-0f643cf0, 0?customtrunk) in new stack -- Executing [...@macro-dialout-trunk:19] Dial(SIP/105-0f643cf0, DAHDI/g1/150|300|) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/150 -- Channel 0/1, span 1 got hangup, cause 44 -- Forcing restart of channel 0/1 on span 1 since channel reported in use -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@macro-dialout-trunk:20] Goto(SIP/105-0f643cf0, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-chanunav...@macro-dialout-trunk:1] GotoIf(SIP/105-0f643cf0, 1?noreport) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) -- Executing [s-chanunav...@macro-dialout-trunk:3] NoOp(SIP/105-0f643cf0, TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 44) - failing through to other trunks) in new stack -- Executing [9...@from-internal:5] Macro(SIP/105-0f643cf0, outisbusy|) in new stack -- Executing [...@macro-outisbusy:1] Playback(SIP/105-0f643cf0, all-circuits-busy-now|noanswer) in new stack -- SIP/105
Re: [asterisk-users] Losing local SIP phones when internet goes down?
Have you tried to set srvlookup=no on your sip.conf? Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Nikhil Nair nn...@pobox.com escreveu: Hi again, OK, I've now installed a local caching nameserver, but don't see any change at all. IN detail, what I did: - Installed Debian packages resolvconf and dnsmasq (resolvconf just takes care of dynamic nameserver allocations in /etc/resolv.conf). - After looking at the docs, edited /etc/network/interfaces to add a dns-nameservers line in the entry for eth1. Then reconfigured resolvconf. - Checked /etc/resolv.conf: now showing 127.0.0.1 as the only nameserver. - Tested name resolution in general: working fine. - Turned ADSL router off and tried to make local and Zap calls: no luck. - Rebooted machine and tried again: still no luck. Again, the logs indicate that Asterisk thinks the SIP phones are unreachable. Was there anything special I needed to do with the setup of dnsmasq, or its interface with Asterisk? If not, I'm stuck again. Thoughts? Nikhil. On Fri, 5 Feb 2010, Nikhil Nair wrote: Hi all, Many thanks for all your very fast and really helpful replies! Now I know about the asynchronous DNS issue, this all starts to make sense: presumably, when I disabled eth1 completely, the DNS queries just failed immediately, so didn't hold anything else up, whereas in the other scenarios (router or phone line down) the DNS requests weren't being answered, so Asterisk was waiting, preventing it from doing other things properly. So, as suggested, I'll go ahead and install dnsmasq (or similar), and that should fix things. Cheers, Nikhil. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
2010/2/5 Vinícius Fontes vinic...@canall.com.br: Have you tried to set srvlookup=no on your sip.conf? I think that just stops SRV lookups, not regular DNS. /r -- _ -- 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] Still on spandsp/app_fax and T.38
On 02/05/2010 07:40 PM, Vinícius Fontes wrote: This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :) Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it might be spandsp-related. What happens is, when I use app_fax to receive a fax using T.38, I get very low transmission rates, and about 30% of the faxes are just not transmitted at all. 2400bps is not uncommon. If instead of T.38 I use an analog landline connected to that very same Asterisk box, app_fax works wonderfully well transmitting at full 9600bps. Just to rule out Asterisk's T.38 handling, I noticed that Fax For Asterisk works perfectly with T.38, also on that same box. Is the FAX machine you are using only capable of 9600bps, because full speed should be 14400bps? Can you get a packet trace of a problem call, using wireshark, and mail it to me? I'll be happy to provide you any info that could help solve this issue. Steve -- _ -- 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] Polycom phone DND state
DND works flawlessly, but whenever using BLF I can only tell that a line is either in use (on a call) or not. I cannot tell a phone is on DND, or on hold for that matter. Would be extremely useful. Would be willing to pay for this developpement if it can be done as long as the feature makes it into trunk. Heck, I'll give 200$ for someone just to tell me how to configure it properly if it's a matter of just missing a config line. Mike Which polycom phones are you using and what SIP firmware are you using? I am using 3.2.0, with a variety of phones (321, 331, 430, 450, 550, 650) Mike -- _ -- 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] Losing local SIP phones when internet goes down?
On 05/02/10 14:21, Nikhil Nair wrote: Hi again, OK, I've now installed a local caching nameserver, but don't see any change at all. IN detail, what I did: - Installed Debian packages resolvconf and dnsmasq (resolvconf just takes care of dynamic nameserver allocations in /etc/resolv.conf). - After looking at the docs, edited /etc/network/interfaces to add a dns-nameservers line in the entry for eth1. Then reconfigured resolvconf. - Checked /etc/resolv.conf: now showing 127.0.0.1 as the only nameserver. - Tested name resolution in general: working fine. - Turned ADSL router off and tried to make local and Zap calls: no luck. - Rebooted machine and tried again: still no luck. Again, the logs indicate that Asterisk thinks the SIP phones are unreachable. I can only indicate what my setup is:- /etc/resolv.conf contains just the real ip of the server 192.168.xxx.xxx /etc/resolv.conf.dnsmasq contains the ip addresses of the dns of my isp it could also have 8.8.8.8 and 8.8.4.4 if you like Google. Dnsmasq is configured to use this file. /etc/hosts on the server contains details of every item on the network and there names. 192.168.nnn.nnn Aastra9133 192.168.nnn.nnn workstation /etc/dhcpd.conf list only 192.168.xxx.xxx as nameserver so with this I can do ping Aastra9133 from any other machine on the network. Hope this helps DC smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Losing local SIP phones when internet goes down?
I solved similar issues by setting srvlookup=no, having bind running locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Randy R randulo2...@gmail.com escreveu: 2010/2/5 Vinícius Fontes vinic...@canall.com.br: Have you tried to set srvlookup=no on your sip.conf? I think that just stops SRV lookups, not regular DNS. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing
Steve Underwood wrote: Spandsp doesn't support those features. I don't know anything which does. It seems they can only be used with TCP. Spandsp does support T38FaxFillBitRemoval which the FAX for Asterisk package does not (according to Commetrex). I added indication of T38FaxTranscodingMMR and T38FaxTranscodingJBIG from app_fax because of the presence of the t38_set_mmr/jbig_transcoding() calls in the spandsp API. I will admit to not reading the documentation to see if they actually did anything useful, though... should I remove them? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
On Fri, 5 Feb 2010, Vinícius Fontes wrote: I solved similar issues by setting srvlookup=no, having bind running locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf. Your local bind is what solved the problem. The srvlookup=no didn't actually help IMO. j Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Randy R randulo2...@gmail.com escreveu: 2010/2/5 Vinícius Fontes vinic...@canall.com.br: Have you tried to set srvlookup=no on your sip.conf? I think that just stops SRV lookups, not regular DNS. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goesdown?
Hello, What do you get with ipconfig from your clients? What do you get with nslookup of a client or server? What do you get with tracert to your server from a client? Can you access the internet from a client? Are you isolated as a private network? Thanks Larry -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Cotton Sent: Friday, February 05, 2010 8:44 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Losing local SIP phones when internet goesdown? On 05/02/10 14:21, Nikhil Nair wrote: Hi again, OK, I've now installed a local caching nameserver, but don't see any change at all. IN detail, what I did: - Installed Debian packages resolvconf and dnsmasq (resolvconf just takes care of dynamic nameserver allocations in /etc/resolv.conf). - After looking at the docs, edited /etc/network/interfaces to add a dns-nameservers line in the entry for eth1. Then reconfigured resolvconf. - Checked /etc/resolv.conf: now showing 127.0.0.1 as the only nameserver. - Tested name resolution in general: working fine. - Turned ADSL router off and tried to make local and Zap calls: no luck. - Rebooted machine and tried again: still no luck. Again, the logs indicate that Asterisk thinks the SIP phones are unreachable. I can only indicate what my setup is:- /etc/resolv.conf contains just the real ip of the server 192.168.xxx.xxx /etc/resolv.conf.dnsmasq contains the ip addresses of the dns of my isp it could also have 8.8.8.8 and 8.8.4.4 if you like Google. Dnsmasq is configured to use this file. /etc/hosts on the server contains details of every item on the network and there names. 192.168.nnn.nnn Aastra9133 192.168.nnn.nnn workstation /etc/dhcpd.conf list only 192.168.xxx.xxx as nameserver so with this I can do ping Aastra9133 from any other machine on the network. Hope this helps DC -- _ -- 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] Losing local SIP phones when internet goes down?
On 05/02/10 16:01, Jeff LaCoursiere wrote: On Fri, 5 Feb 2010, Vinícius Fontes wrote: I solved similar issues by setting srvlookup=no, having bind running locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf. Your local bind is what solved the problem. The srvlookup=no didn't actually help IMO. Given the choice between configuring bind and dnsmasq I know which I'd go for. DC smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Still on spandsp/app_fax and T.38
Yes, the fax machine only transmits at 9600. That's normal and expected. I'll capture the packets and will provide you with a link to download it in a few minutes. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Steve Underwood ste...@coppice.org escreveu: On 02/05/2010 07:40 PM, Vinícius Fontes wrote: This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :) Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it might be spandsp-related. What happens is, when I use app_fax to receive a fax using T.38, I get very low transmission rates, and about 30% of the faxes are just not transmitted at all. 2400bps is not uncommon. If instead of T.38 I use an analog landline connected to that very same Asterisk box, app_fax works wonderfully well transmitting at full 9600bps. Just to rule out Asterisk's T.38 handling, I noticed that Fax For Asterisk works perfectly with T.38, also on that same box. Is the FAX machine you are using only capable of 9600bps, because full speed should be 14400bps? Can you get a packet trace of a problem call, using wireshark, and mail it to me? I'll be happy to provide you any info that could help solve this issue. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
Could be. Important thing is the problem was solved :) Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Jeff LaCoursiere j...@jeff.net escreveu: On Fri, 5 Feb 2010, Vinícius Fontes wrote: I solved similar issues by setting srvlookup=no, having bind running locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf. Your local bind is what solved the problem. The srvlookup=no didn't actually help IMO. j Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Randy R randulo2...@gmail.com escreveu: 2010/2/5 Vinícius Fontes vinic...@canall.com.br: Have you tried to set srvlookup=no on your sip.conf? I think that just stops SRV lookups, not regular DNS. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Still on spandsp/app_fax and T.38
Here's the packet trace I promised: http://www.zshare.net/download/72186098494e6f8c/. As this is a production system, there were a few calls along with the one that interests us. The one you're looking for is that from 5433142...@10.150.65.16 to 5421047...@10.153.66.146. The provider has the address 10.150.65.16 and my box has the address 10.153.66.146. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Vinícius Fontes vinic...@canall.com.br escreveu: Yes, the fax machine only transmits at 9600. That's normal and expected. I'll capture the packets and will provide you with a link to download it in a few minutes. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Steve Underwood ste...@coppice.org escreveu: On 02/05/2010 07:40 PM, Vinícius Fontes wrote: This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :) Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it might be spandsp-related. What happens is, when I use app_fax to receive a fax using T.38, I get very low transmission rates, and about 30% of the faxes are just not transmitted at all. 2400bps is not uncommon. If instead of T.38 I use an analog landline connected to that very same Asterisk box, app_fax works wonderfully well transmitting at full 9600bps. Just to rule out Asterisk's T.38 handling, I noticed that Fax For Asterisk works perfectly with T.38, also on that same box. Is the FAX machine you are using only capable of 9600bps, because full speed should be 14400bps? Can you get a packet trace of a problem call, using wireshark, and mail it to me? I'll be happy to provide you any info that could help solve this issue. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 Asterisk Boxes, Single Voicemail
Searching through the archives, I couldn't find an answer for this... I have two asterisk systems, (system A and system B), and would like to use a single voicemail system. Phones on system B are SIP phones, registered at system B. Can the message-waiting indicator be activated on a SIP phone registered to system B, if the voicemail resides on system A? If so, how? Thanks, folks. -- _ -- 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] 2 Asterisk Boxes, Single Voicemail
Speaking just for polycom 501's (3 lines), you can set MWI on two different systems by registering a line to each system; for your question, I'd do it like this; Exten 100 is on system B and sends it's VM to exten 110 on System A. You register line 1 as 100 on system b and line 2 as 110 on System A. when you leave a voicemail for 100, the 110 message will light up MWI. Regards, Danny Nicholas -- _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Greg Blakely Sent: Friday, February 05, 2010 10:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 2 Asterisk Boxes, Single Voicemail Searching through the archives, I couldn't find an answer for this... I have two asterisk systems, (system A and system B), and would like to use a single voicemail system. Phones on system B are SIP phones, registered at system B. Can the message-waiting indicator be activated on a SIP phone registered to system B, if the voicemail resides on system A? If so, how? Thanks, folks. -- _ -- 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] Losing local SIP phones when internet goes down?
OK, I've now installed a local caching nameserver, but don't see any change at all. - Tested name resolution in general: working fine. - Turned ADSL router off and tried to make local and Zap calls: no luck. Did you try to make any calls before pulling the plug on the ADSL router? dnsmasq wouldn't have had the chance to cache any lookups that are made if you didn't. - Rebooted machine and tried again: still no luck. Again, the logs indicate that Asterisk thinks the SIP phones are unreachable. Was there anything special I needed to do with the setup of dnsmasq, or its interface with Asterisk? If not, I'm stuck again. Turn on logging of dns queries in dnsmasq. (-q or --log-queries). That should 1) verify that it is actually being used for lookups 2) tell you what, exactly, it is being asked to resolve. (which you could then add to your hosts file) -- _ -- 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] 2 Asterisk Boxes, Single Voicemail
On 5 Feb 2010, at 16:55, Greg Blakely wrote: If so, how? NFS or rsync? S -- _ -- 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] 2 Asterisk Boxes, Single Voicemail
On 5 Feb 2010, at 16:55, Greg Blakely wrote: If so, how? NFS or rsync? S Use ODBC voice message storage and realtime voicemail configuration. - Sean -- _ -- 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] Still on spandsp/app_fax and T.38
There you go: http://www.canall.com.br/wireshark_trace_t38_ffa.gz Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Steve Underwood ste...@coppice.org escreveu: Hi Vinícius, Asterisk + Spandsp is working correctly in that call. The other system sends bad training data at V.29/9600bps. Spandsp rejects it. The other system sends bad training data at V.27ter/4800bps. Spandsp rejects it. The other system sends clean training data at V.27ter/2400bps. Spandsp accepts it. The other system sends a page at V.27ter/2400bps. Spandsp accepts it. The bad training data is *really* bad. It should be 1.5s of all zero bits. It starts off with zeros, but after a few hundred milliseconds it changes to complete rubbish. I can't believe the Commetrex engine in Digium's FAX for Asterisk would accept this. Perhaps something subtle means they are sent the correct data. Can you send a wireshark log of a call with FAX for Asterisk? Steve On 02/06/2010 12:43 AM, Vinícius Fontes wrote: No problem, hosted it on my company's website: http://www.canall.com.br/wireshark_trace_t38.gz. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Steve Underwoodste...@coppice.org escreveu: On 02/06/2010 12:01 AM, Vinícius Fontes wrote: Here's the packet trace I promised: http://www.zshare.net/download/72186098494e6f8c/. As this is a production system, there were a few calls along with the one that interests us. The one you're looking for is that from 5433142...@10.150.65.16 to 5421047...@10.153.66.146. The provider has the address 10.150.65.16 and my box has the address 10.153.66.146. Can you put the file somewhere that actually works. I've downloaded it 5 times now, and it has been cutoff at different points each time. These free file sharing services all seem to do this. Maybe they all run the same broken software. Steve -- _ -- 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] SS7 and Asterisk
Please some one shed some light on it.. On Thu, Feb 4, 2010 at 6:48 PM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Hello All, Please let me know Answers to the following questions .Backgroud. 1. Which one is better to use libss7 or chan_ss7. Today first time i come to know about it ... little bit i googled but need experts comment on it. 2. I have perpared my server ie installed Asterisk , Configured TE420P delevired to Telco the Operator. I come to know that ss7 signalling will be used. Now can i install SS7 signalling as Asterisk and other things are already installed OR i may do the installation from scratch then install in sequence. Please keep in mind i heard this thing today ... made search for it about 3 hours.. It will be really nice if some one shed some light on its configuration Please shed some light on it Thanks Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas -- _ -- 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] SS7 and Asterisk
- ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Please some one shed some light on it.. Sangoma offers SS7 solutions for Asterisk using their SMG platform: http://sangoma.com/products/software_building_blocks/ss7_solutions/ss7_signaling_and_media_gateway.html --Tim -- _ -- 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] 1.6.2.1: DTMF trouble with PSTN
sean darcy wrote: Using 1.6.2.1 with a TDM400, attached to internal analog phones and PSTN. When I dial out to PSTN, I cannot send tones, like push 1 for something stupid. The call itself works, but the DTMF tones fail. -- Starting simple switch on 'DAHDI/1-1' -- Executing [6258...@internal:1] Answer(DAHDI/1-1, ) in new stack -- Executing [6258...@internal:2] Dial(DAHDI/1-1, DAHDI/4/ww2156258013) in new stack -- Called 4/ww2156258013 -- DAHDI/4-1 answered DAHDI/1-1 -- Native bridging DAHDI/1-1 and DAHDI/4-1 -- Hungup 'DAHDI/4-1' Any suggestions? sean This is DAHDI Tools Version - 2.2.1 Do DTMF tones work for others over dahdi? I'd file a bug, but I'd like to make sure it's not just my mistake. sean -- _ -- 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] Sipgate.co.uk on Asterisk 1.6.2.2
Hi all, I have been running Asterisk for years (CVS-HEAD on 2005-08-24) with no problems save a failed harddrive. I have decided to build a new box and have Asterisk 1.6.2.2 playing nicely with mISDN after lots of changes to dialplan syntax etc. I am struggling with SIP trunks to sipgate.co.uk and dualtalk.com. Does anyone have a working examples? When I make an outgoing call I get... [Feb 5 21:15:47] WARNING[5994]: chan_sip.c:5329 create_addr: No such host: sipgate2 [Feb 5 21:15:47] WARNING[5994]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) Sip show registry implies everything is well... Host dnsmgr Username Refresh State Reg.Time sipgate.co.uk:5060 N 222105 Registered Fri, 05 Feb 2010 21:06:47 sipgate.co.uk:5060 N 111105 Registered Fri, 05 Feb 2010 21:06:47 sip.dualtalk.com:5060 N 31785 Registered Fri, 05 Feb 2010 21:06:47 Register section of my sip.conf... Register = 3:444...@sip.dualtalk.com 3%3a444...@sip.dualtalk.com Register = 111: 1secr...@sipgate.co.uk/111 Register = 222: 2secr...@sipgate.co.uk/222 One of the sipgate sections of my sip.conf... [Sipgate2] type=friend username=222 secret=2secret2 host=sipgate.co.uk fromuser=222 fromdomain=sipgate.co.uk nat=yes authuser=222 dtmfmode=rfc2833 context=sipgate_ic insecure=very canreinvite=no disallow=all allow=alaw Thanks in advance! -- _ -- 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] large scale paging
Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097 -- _ -- 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] large scale paging
Mark Willis wrote: This could potentially create a very weird audio situation where the delay between adjacent phones is audible so instead of acting like loudspeakers in parallel on a conventional system, it just sounds like a bunch of people talking at once and is not understandable. Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097 -- _ -- 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] large scale paging
Mark Willis schrieb: Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Multicast RTP might be the solution. http://wiki.snom.com/Settings/multicast_listen http://wiki.snom.com/Settings/multicast_address http://forum.snom.com/index.php?showtopic=1905 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- -- _ -- 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] large scale paging
On Fri, 5 Feb 2010, Mark Willis wrote: Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. How did you get that number? Even with ulaw @ 64Kbps you theoretically get 32Mbps. If you used G.729 you would cut that down to 4 or 5Mbps. Totally oversimplified, but that seems a lot more doable. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? I can't say I have ever pushed that hard, but that doesn't sound like it would be difficult to handle. There are plenty claiming they have 400 simultaneous two way conversations going on a single box. j Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] large scale paging
On Fri, Feb 5, 2010 at 4:50 PM, Mark Willis marksli...@markwillis.net wrote: Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097 What you really want is multicast RTP, preferably as implemented in SNOM phones: https://issues.asterisk.org/view.php?id=11797 http://wiki.snom.com/Settings/multicast_address One RTP stream, any number of receivers, no SIP session. Doing this with unicast RTP and individual INVITEs would be tough. If your system can't do 500 call setups per second (or better) you'll introduce massive delays in call setup to the recipients, not to mention serious RTP burden with that many streams. I hope you haven't bought phones yet (or bought Snom) ;). -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] large scale paging
I thought of that too, but the phones will be spread over a large number of rooms in several buildings, so that won't be too much of an issue. Mark Willis On 2010-02-05 15:55, jon pounder wrote: Mark Willis wrote: This could potentially create a very weird audio situation where the delay between adjacent phones is audible so instead of acting like loudspeakers in parallel on a conventional system, it just sounds like a bunch of people talking at once and is not understandable. Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097 -- _ -- 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] large scale paging
Hi! Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot Use multi-cast: Read the See also section at the bottom of this page and look at MAST and/or app_rtppage. A couple of phone vendors have multi-cast support in their models. http://www.voip-info.org/wiki/view/Asterisk+cmd+Page http://www.voip- info.org/wiki/index.php?page=Asterisk+phone+snom#RelatedMulticastapp_rtppa geAsterisk16orl Philipp -- _ -- 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] Asterisk going down
Hello my friends, My asterisk is going down randomly, following you will find some errors that i could see in the /var/log/asterisk/message at the moment of the crash: [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Maximum retries exceeded on transmission 1850202...@10.4.1.152 for seqno 21 (Critical Response) [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Hanging up call 1850202...@10.4.1.152 - no reply to our critical packet. [Feb 5 10:33:04] NOTICE[6519] chan_sip.c: Call from '346' to extension '3415554' rejected because extension not found. [Feb 5 10:35:31] NOTICE[6519] chan_sip.c: Disconnecting call 'SIP/301-09ad3be8' for lack of RTP activity in 301 seconds [Feb 5 10:36:17] NOTICE[6519] chan_sip.c: Disconnecting call 'SIP/317-b7735220' for lack of RTP activity in 301 seconds [Feb 5 10:38:19] NOTICE[6519] chan_sip.c: Peer '353' is now Reachable. (1ms / 2000ms) [Feb 5 10:42:59] NOTICE[6519] chan_sip.c: Peer '353' is now Reachable. (7ms / 2000ms) [Feb 5 10:51:09] NOTICE[6519] chan_sip.c: Peer '358' is now Reachable. (1ms / 2000ms) [Feb 5 10:53:08] NOTICE[6519] chan_sip.c: Peer '366' is now UNREACHABLE! Last qualify: 108 But later, at 2 pm, Asterisk went down again but with no weird message in /var/log/asterisk/message (just some unreachable messages of some extensions that has always been in the console since i installed Asterisk, but it never crash Asterisk untill last weeks ago): [Feb 5 13:54:11] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Feb 5 13:55:18] NOTICE[6536] chan_sip.c: Registration from ' sip:3...@10.4.1.6:5060' failed for '10.4.2.3' - No matching peer found [Feb 5 13:57:40] NOTICE[6536] chan_sip.c: Call from '346' to extension '04265417457' rejected because extension not found. [Feb 5 13:59:15] NOTICE[6536] chan_sip.c: Peer '341' is now Reachable. (2ms / 2000ms) [Feb 5 13:59:25] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Feb 5 14:01:43] NOTICE[6536] chan_sip.c: Peer '339' is now UNREACHABLE! Last qualify: 101 [Feb 5 14:04:22] NOTICE[6536] chan_sip.c: Peer '339' is now Reachable. (44ms / 2000ms) [Feb 5 14:04:39] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Feb 5 14:09:53] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) I could not make any call, neither internall nor to the pstn, what could be happening here my friends? what should i check in the Asterisk server? is this a network problem? memmory or cpu problems? Thanks in advance for your help -- _ -- 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] strange issue with iptables + Asterisk
Hi all, I'm having a strange issue, wanted to see if anyone had any suggestions. Due to the recent spike in VoIP related hacking attempts I decided to tighten security by writing iptables scripts to only allow traffic to my servers which is white-listed, since then I've had an issue under certain circumstances. I have two boxes (gateway) + (end-point), both running Asterisk 1.4.29 and connecting to each other via IAX2. They are able to call each other just fine. The (gateway) box connects to providers for access to PSTN via SIP. After hours, if you dial the (end-point) server through the PSTN (aka, it flows through PSTN - gateway - end-point) the behavior of the system is to take the call and forward it to an outside DID, the call goes back out through the gateway and to PSTN. This works perfectly with iptables filters on the gateway box turned off, when they are on I get no audio. Meanwhile, all other calls in and out work perfectly. I did a packet capture from gateway - end-point and found all the IAX2 signaling packets there but no media packets (aka no audio). Then I discovered that if I put a 3 second pause on the end-point box before forwarding the call, the audio is passed on to PSTN and the problem solved. Again, if I turn iptables off on the gateway machine everything works without the delay. The immediate issue is solved but I'd like to know if anyone seen anything like this before, it may cause problems for people trying to tighten security.. This is the iptables script: http://bash.pastebin.com/m39babd2b -- Ernesto smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] large scale paging
Thanks everyone, I'll look at multicast. The customer prefers Snom phones, luckily. Mark On 2010-02-05 16:32, Philipp von Klitzing wrote: Hi! Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot Use multi-cast: Read the See also section at the bottom of this page and look at MAST and/or app_rtppage. A couple of phone vendors have multi-cast support in their models. http://www.voip-info.org/wiki/view/Asterisk+cmd+Page http://www.voip- info.org/wiki/index.php?page=Asterisk+phone+snom#RelatedMulticastapp_rtppa geAsterisk16orl Philipp -- _ -- 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] large scale paging
On 2010-02-05 16:20, Jeff LaCoursiere wrote: On Fri, 5 Feb 2010, Mark Willis wrote: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. How did you get that number? Even with ulaw @ 64Kbps you theoretically get 32Mbps. If you used G.729 you would cut that down to 4 or 5Mbps. Totally oversimplified, but that seems a lot more doable. UDP ethernet overhead. I used 87kbps. This site says something similar, although they get there a different way: http://site.asteriskguide.com/bandcalc/bandcalc.php I expect the INVITEs and buffering of 500 streams will be the real problem, which is why I'm thinking that the people who suggested multicast are right. Mark -- _ -- 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] Recording Calls
Hello everyone I have a central Avaya S8300 with G450 Gateway, now all calls go through the Avaya, but I need to record all calls, my questions are: 1- Can I to interconnect Asterisk with Avaya ? 2- With that tool might Asterisk record calls. I hope your suggestions. Thanks Greetings, _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] Dial script
Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
On Fri, 5 Feb 2010, Thomas Perron wrote: Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. Do you mean the dialed numbers are in sequence like 555-555-0001, 555-555-0002* or do you mean dialing the numbers one after the other from a list of customers you have a pre-existing business relationship with? I'm guessing you don't want to sit there from start to finish :) You'll need some sort of database to keep track of which numbers have been called and where to start the next time. You could write a program to create call files or you could write a program to connect to your Asterisk server using AMI and issue originate commands. *) Probably illegal in the United States and any other civilized country. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
Try this: #rm -rf / - Original Message - From: Thomas Perron thomas.per...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, February 05, 2010 8:54 PM Subject: [asterisk-users] Dial script Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Dial script
karl, does it make you feel good ? wow. pathetic. On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife karlf...@gmail.com wrote: Try this: #rm -rf / - Original Message - From: Thomas Perron thomas.per...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, February 05, 2010 8:54 PM Subject: [asterisk-users] Dial script Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
On Fri, 5 Feb 2010, Karl Fife wrote: Try this: #rm -rf / Copycat! On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20 Sure. Add this to root's crontab: * * * * rm --farce --recursive / -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
On Sat, 6 Feb 2010, Thomas Perron wrote: karl, does it make you feel good ? wow. pathetic. On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife karlf...@gmail.com wrote: Try this: #rm -rf / I second that opinion. Tell us first WHY you want to dial 1 numbers in sequence. Without any reason, you must be assumed to be a call spammer, and you are looking for help in the wrong place. j - Original Message - From: Thomas Perron thomas.per...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, February 05, 2010 8:54 PM Subject: [asterisk-users] Dial script Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
I think he's referring to the fact that you seem to be looking to put together the telephone equivalent of a spam service. I'd be advising rm -rf / as well. On 02/06/10 16:19, Thomas Perron wrote: karl, does it make you feel good ? wow. pathetic. On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife karlf...@gmail.com wrote: Try this: #rm -rf / - Original Message - From: Thomas Perron thomas.per...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, February 05, 2010 8:54 PM Subject: [asterisk-users] Dial script Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
- Original Message - From: Thomas Perron thomas.per...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, February 05, 2010 8:54 PM Subject: [asterisk-users] Dial script Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife karlf...@gmail.com wrote: Try this: #rm -rf / From: Thomas Perron thomas.per...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk- us...@lists.digium.com Sent: Friday, February 05, 2010 11:19 PM Subject: Re: [asterisk-users] Dial script karl, does it make you feel good ? wow. pathetic. Yes sir, yes it does. I encourage you to make an ass of me for assuming the worst: Please enlighten me and everyone on this list as to the legitimate, curteous practice you are engaging in that requires you to serially dial 10,000 numbers, awaiting human response. Your original query was NOT how can I draw from a 10,000 row database of opt-in's Otherwise yes. Giving a war-dialer something like a unix-finger-gesture feels pretty good. What would make me feel even BETTER would be to be proven wrong for assuming the worst about a member of this community. -Karl - Original Message - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
Nice. :-) Didn't see that, I concede. - Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 12:10 AM Subject: Re: [asterisk-users] Dial script On Fri, 5 Feb 2010, Karl Fife wrote: Try this: #rm -rf / Copycat! On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20 Sure. Add this to root's crontab: * * * * rm --farce --recursive / -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
On Fri, Feb 05, 2010 at 01:21:38PM +, Nikhil Nair wrote: Hi again, OK, I've now installed a local caching nameserver, but don't see any change at all. Just to add to the discussion, my setup I was using a local bind9 server for local/authorative and recursive queries I think from memory it was asking for a sip address to register with and the record had a ttl or 600 (5min) so could expire very easily. can I suggest maybe whilst eth1 (the internet link) is down, stop and restart asterisk with logging and check to see what fails IN detail, what I did: [snip] signature.asc Description: Digital signature -- _ -- 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] Website Down ?
Hi, Have I missed something as http://downloads.asterisk.org is not available ? -- Thanks, Phil -- _ -- 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] Website Down ?
Down for me too. -K - Original Message - From: --[ UxBoD ]-- ux...@splatnix.net To: asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 12:48 AM Subject: [asterisk-users] Website Down ? Hi, Have I missed something as http://downloads.asterisk.org is not available ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TOS bits, DSCP, Asterisk Polycom
Has anyone figured this out yet? Lots of places say to add the following to sip.conf of an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ; Sets TOS for RTP video packets. To match the current 1.2 machine would I set the Polycom's sip.cfg to the first or second QOS option? Option 1: ~~~ QOS Ethernet RTP qos.ethernet.rtp.user_priority=5/ CallControl qos.ethernet.callControl.user_priority=5/ Other qos.ethernet.other.user_priority=2/ /Ethernet IP RTP qos.ip.rtp.dscp= qos.ip.rtp.min_delay=1 qos.ip.rtp.max_throughput=1 qos.ip.rtp.max_reliability=1 qos.ip.rtp.min_cost=0 qos.ip.rtp.precedence=5/ CallControl qos.ip.callControl.dscp= qos.ip.callControl.min_delay=1 qos.ip.callControl.max_throughput=1 qos.ip.callControl.max_reliability=1 qos.ip.callControl.min_cost=0 qos.ip.callControl.precedence=5/ /IP /QOS ~~~ Option 2: ~~~ QOS Ethernet RTP qos.ethernet.rtp.user_priority=5/ CallControl qos.ethernet.callControl.user_priority=5/ Other qos.ethernet.other.user_priority=2/ /Ethernet IP RTP qos.ip.rtp.dscp=ef qos.ip.rtp.min_delay=1 qos.ip.rtp.max_throughput=1 qos.ip.rtp.max_reliability=1 qos.ip.rtp.min_cost=0 qos.ip.rtp.precedence=5/ CallControl qos.ip.callControl.dscp=ef qos.ip.callControl.min_delay=1 qos.ip.callControl.max_throughput=1 qos.ip.callControl.max_reliability=1 qos.ip.callControl.min_cost=0 qos.ip.callControl.precedence=5/ /IP /QOS ~~~ or none of the above? Also, how does 10111000 Fit into: [ 0 1 2 ] [3] [4] [5] [6 7] [ Precedence ] [D] [T] [R] [ECN Field] Is it read backwards? Any helpful comments appreciated. References: http://en.wikipedia.org/wiki/Type_of_Service#Type_of_Service http://en.wikipedia.org/wiki/DiffServ#Expedited_Forwarding_.28EF.29_PHB_-_DSCP.3D.2846_OR_101110.29 http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+tos http://www.polycom.com/global/documents/support/setup_maintenance/products/voice/SoundPointIP_SoundStationIP_AdminGuide_SIP3_0_Eng_Rev_A.pdf -- _ -- 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
