[asterisk-users] sip pregi net account registration
Hi guys, I am trying to configure sip.pregi.net account with my Asterisk 1.4.X, since its a free account, its not getting registered, even my machine IP is allowed in firewall. In the same machine if i register openser account which is in public i am able to register. while checking the sip debug the register request is keep on sending but there is no response. what i did is i registered the same account in my softphone installed in my laptop, there it got registered. only with Asterisk its not registering, I tried allowing externip as my routers IP, even then its not getting registered. Does anyone used sip.pregi.net account with Asterisk? If so let me know the settings. Thank you. -- _ -- 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 CDRs
I am using AMI Originate on asterisk 1.8.11 on SIP channels. I have set unanswered=yes in cdr.conf because I want to log NO ANSWER and BUSY calls. The issue is, that if a SIP peer is not registered, and an originate request is made for that peer, a null cdr entry is made as follows: ,2012-04-05 09:28:53,,2012-04-05 09:28:53,0,0,FAILED,DOCUMENTATION,, How can I fix it? Or, how can I set cdr not to log entries if a channel doesn't exist. -- _ -- 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] issue with Digium TDM410P
On Thu, Apr 05, 2012 at 08:14:09AM -0400, Mathieu Therrien wrote: Thanks for answer. Devices PAP2 and SPA3000 works good and use FSK for VMWI and CallerID, so I think TDM410P should also works. CallerID works good. So issue should be in source code of VMWI. Alec, If FSK is used for VMWI on his working setup, this doesn't sound like it would be fixed by your Generate VMWI neon pulses from FXS module.. patch to the wctdm24xxp DAHDI driver [1] I'm assuming. Must be something in chan_dahdi? But Now that the changes to the wctdm24xxp have calmed down (for awhile now... :/) it looks like it might be time to try again to rebase those patches on the current trunk. [1] https://reviewboard.asterisk.org/r/1144/ Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] issue with Digium TDM410P
I don't think neon use FSK, neon should use high voltage. So I think, the issue should be on FSK in DAHDI module code. On Thu, Apr 5, 2012 at 11:50, Shaun Ruffell sruff...@digium.com wrote: On Thu, Apr 05, 2012 at 08:14:09AM -0400, Mathieu Therrien wrote: Thanks for answer. Devices PAP2 and SPA3000 works good and use FSK for VMWI and CallerID, so I think TDM410P should also works. CallerID works good. So issue should be in source code of VMWI. Alec, If FSK is used for VMWI on his working setup, this doesn't sound like it would be fixed by your Generate VMWI neon pulses from FXS module.. patch to the wctdm24xxp DAHDI driver [1] I'm assuming. Must be something in chan_dahdi? But Now that the changes to the wctdm24xxp have calmed down (for awhile now... :/) it looks like it might be time to try again to rebase those patches on the current trunk. [1] https://reviewboard.asterisk.org/r/1144/ Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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 -- Mathieu Therrien, VE2TMQ / VA2IO B. Sc. A. Genie Logiciel www.ve2tmq.ca VoIP: sip:*99...@sipbroker.com iNUM: +883-5100-099-01841 PSTN: +1-514-316-9498 PGP id : 0xA2882612 PGP FingerPrint : FC7C 9D8A 11BE 7C0E C95B 1D2A E372 338C A288 2612 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial Plan - Routing via Caller ID
I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 614000 DID2 = 614001 CNAME1 = 614999 CNAME2 = 614998 CNAME3 = 614997 context1 context2 context3 I have looked at several examples (patterns) and I am a little confused and am looking to the list for guidance. Assuming I have (2) two DIDs and I want to route incoming calls according to the DID called and as well as the Caller ID various contexts. If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) I have seen examples where I could use a pattern like (not specifying a Caller ID info, and that works fine): exten = _X!,n,Goto(context1,s,1) exten = _X!,n,Goto(context2,s,1) exten = _X!,n,Goto(context3,s,1) I am confused on how to use patterns. I would like to learn how I can take either DID and route the calls to various contexts via the CallerID (which couild be the entire DID number, an NPA only or an NPANXX. Could someone please give me an example on how to do this? TIA -- _ -- 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] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 10:35 AM, list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) You are pretty close, I think it's your priorities that are the problem. When I use Ex Girl Friend Logic I write my extensions this way: exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000,1,Dial(SIP/,25) I have never used a wildcard match like you are attempting to do with the 702 prefix but according to voip-info.org it should work -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 10:52 AM, John Kiniston johnkinis...@gmail.comwrote: On Thu, Apr 5, 2012 at 10:35 AM, list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) You are pretty close, I think it's your priorities that are the problem. When I use Ex Girl Friend Logic I write my extensions this way: exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000,1,Dial(SIP/,25) I agree, priorities are very tricky in this case, and I've spent a lot of time figuring out similar scenarios. Also you may need to use an 's' priority in some cases, where there are two potential matches that are the same priority. I'm sorry I can't think of where I have a useful code snippet for your exact case. I'd recommend starting with a fully working explicit statement then work back from there to less explicit: exten = 16027170050/8774613644,1,Goto(monkeys,s,1) Then: exten = 1602717005X/8774613644,1,Goto(monkeys,s,1) Etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Dial Plan - Routing via Caller ID
Priorities are not complicated. Each extension starts with priority 1, all additional priorities are n and you ALWAYS end your extension with a Goto or a Hangup so the call doesn't fall off your intended extension and hump into the middle of some other extension and screw everything up. You are pretty close, I think it's your priorities that are the problem. When I use Ex Girl Friend Logic I write my extensions this way: exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000,1,Dial(SIP/,25) I agree, priorities are very tricky in this case, and I've spent a lot of time figuring out similar scenarios. Also you may need to use an 's' priority in some cases, where there are two potential matches that are the same priority. I'm sorry I can't think of where I have a useful code snippet for your exact case. I'd recommend starting with a fully working explicit statement then work back from there to less explicit: exten = 16027170050/8774613644,1,Goto(monkeys,s,1) Then: exten = 1602717005X/8774613644,1,Goto(monkeys,s,1) Etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Dial Plan - Routing via Caller ID
On Thu, 5 Apr 2012 13:35:51 -0400 list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) I think the n's should all be 1's, like so: exten = 614000/_702XXX,1,Goto(context1,s,1) exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000/614997,1,Dial(SIP/,25) The 'n' priority is used for subsequent lines (after the first) in the same extension, but the first one for each extension should be 1. I have seen examples where I could use a pattern like (not specifying a Caller ID info, and that works fine): exten = _X!,n,Goto(context1,s,1) exten = _X!,n,Goto(context2,s,1) exten = _X!,n,Goto(context3,s,1) I suggest you don't use _X! or _X. as a pattern, until you fully understand the security risks. In the asterisk-1.8 tarball, there's a file called README-SERIOUSLY.bestpractices.txt that explains it all. You should read that before you do anything. I am confused on how to use patterns. I would like to learn how I can take either DID and route the calls to various contexts via the CallerID (which couild be the entire DID number, an NPA only or an NPANXX. You have an example for NPA only in the line that handles area code 702. Similar for NPANXX: exten = 614000/_614999,1,Goto(context,s,1) This is all covered quite well on the voip-info wiki: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] issue with Digium TDM410P
On Thu, Apr 05, 2012 at 01:07:11PM -0400, Mathieu Therrien wrote: I don't think neon use FSK, neon should use high voltage. So I think, the issue should be on FSK in DAHDI module code. It sounds like we're in agreement. Although, to be clear, DAHDI-Linux (the kernel modules) do not generate or decode FSK spills. That is done by chan_dahdi in Asterisk. On Thu, Apr 5, 2012 at 11:50, Shaun Ruffell sruff...@digium.com wrote: On Thu, Apr 05, 2012 at 08:14:09AM -0400, Mathieu Therrien wrote: Thanks for answer. Devices PAP2 and SPA3000 works good and use FSK for VMWI and CallerID, so I think TDM410P should also works. CallerID works good. So issue should be in source code of VMWI. Alec, If FSK is used for VMWI on his working setup, this doesn't sound like it would be fixed by your Generate VMWI neon pulses from FXS module.. patch to the wctdm24xxp DAHDI driver [1] I'm assuming. Must be something in chan_dahdi? But Now that the changes to the wctdm24xxp have calmed down (for awhile now... :/) it looks like it might be time to try again to rebase those patches on the current trunk. [1] https://reviewboard.asterisk.org/r/1144/ Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 11:00 AM, Eric Wieling ewiel...@nyigc.com wrote: Priorities are not complicated. Each extension starts with priority 1, all additional priorities are n and you ALWAYS end your extension with a This isn't correct, there are many cases where you must use an 's' priority. Our system simply couldn't function without it. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 pregi net account registration
On Apr 5, 2012, at 3:21 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi guys, I am trying to configure sip.pregi.net account with my Asterisk 1.4.X, since its a free account, its not getting registered, even my machine IP is allowed in firewall. In the same machine if i register openser account which is in public i am able to register. while checking the sip debug the register request is keep on sending but there is no response. what i did is i registered the same account in my softphone installed in my laptop, there it got registered. only with Asterisk its not registering, I tried allowing externip as my routers IP, even then its not getting registered. What settings are you currently using, and what does your infrastructure look like? -- Thanks, Warren Selby, dCAP-- _ -- 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] Dial Plan - Routing via Caller ID
On Apr 5, 2012, at 1:23 PM, Carlos Alvarez car...@televolve.com wrote: On Thu, Apr 5, 2012 at 11:00 AM, Eric Wieling ewiel...@nyigc.com wrote: Priorities are not complicated. Each extension starts with priority 1, all additional priorities are n and you ALWAYS end your extension with a This isn't correct, there are many cases where you must use an 's' priority. Our system simply couldn't function without it. You're think of EXTENSION 's', not PRORITY. Priority is the order the call goes through the dial plan. Extension is the part of the dial plan you're traversing. Priority will always be either a number or an 'n'. exten = EXTENSION,PRIORITY,COMMAND -- Thanks, Warren Selby, dCAP-- _ -- 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] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 12:11 PM, Eric Wieling ewiel...@nyigc.com wrote:
Are you sure you are not referring to the s extension?
Absolutely. Every time I discuss 's' priority on this list or the Asterisk
IRC channel people tell me it either doesn't exist or is wrong, but it's a
powerful under-utilized feature. It's at the core of initially routing
calls on our system.
Show an example of needing s as a priority.
This is from our system, the asterisks have been used to obscure for
privacy, they are numbers.
exten = 1602889,n,Goto(starnetworks#main|s|1)
exten = 1602400,s,Goto(starnetworks#extensions,9520,1)
exten = 1480241,s,Goto(starnetworks#extensions,9766,1)
exten = _X.,s,Goto(starnetworks#extensions|${EXTEN:7}|1)
--
Carlos Alvarez
TelEvolve
602-889-3003
--
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Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 12:13 PM, Warren Selby wcse...@selbytech.com wrote: You're think of EXTENSION 's', not PRORITY. Priority is the order the call goes through the dial plan. Extension is the part of the dial plan you're traversing. Priority will always be either a number or an 'n'. exten = EXTENSION,PRIORITY,COMMAND Nope, it's 's' priority. See my subsequent message about it. Priority CAN be an 's' and in certain situations MUST be an 's' to function. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Apr 5, 2012, at 2:32 PM, Carlos Alvarez car...@televolve.com wrote:
On Thu, Apr 5, 2012 at 12:11 PM, Eric Wieling ewiel...@nyigc.com wrote:
Are you sure you are not referring to the s extension?
Absolutely. Every time I discuss 's' priority on this list or the Asterisk
IRC channel people tell me it either doesn't exist or is wrong, but it's a
powerful under-utilized feature. It's at the core of initially routing calls
on our system.
Show an example of needing s as a priority.
This is from our system, the asterisks have been used to obscure for privacy,
they are numbers.
exten = 1602889,n,Goto(starnetworks#main|s|1)
exten = 1602400,s,Goto(starnetworks#extensions,9520,1)
exten = 1480241,s,Goto(starnetworks#extensions,9766,1)
exten = _X.,s,Goto(starnetworks#extensions|${EXTEN:7}|1)
I still don't understand what you would need this for. What version of asterisk
are you using? From voip-info.org, it says the s priority is used when
different patterns may match at the same point in the extension and act
differently for them, but couldn't you basically do the same thing with
priority labels? How would you ever end up with different patterns matching at
the same point in an extension? Where is your priority 1?
--
Thanks,
Warren Selby, dCAP--
_
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Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 12:57 PM, Warren Selby wcse...@selbytech.com wrote: I still don't understand what you would need this for. What version of asterisk are you using? From voip-info.org, it says the s priority is used when different patterns may match at the same point in the extension and act differently for them, but couldn't you basically do the same thing with priority labels? How would you ever end up with different patterns matching at the same point in an extension? Where is your priority 1? Well, now we get into a lot of design philosophy discussion that I really don't have time for today. I will note that Kevin Fleming wrote the 's' feature into the code before he worked at Digium and still owned the company I now run... I didn't understand his design at all for a long time, but now it's second nature. The 1 priority is in another context that pre-processes the calls, then each customer has a series of 's' priority lines for their individual DID numbers. I use the 's' priority in some DNIS/CID-based call blocking here and there. As far as versions, 1.2, 1.4, and 1.6. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] issue with Digium TDM410P
Sorry, I didn't word my reply correctly, I wasn't trying to say 'hvac' support on the TDM410P would fix the issue. As I think about it more, having the option enabled there doesn't matter, even for a TDM410P. Shaun: Regarding neon patch (review 1144) for wctdm24xxp, I'll have to get on to that. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Friday, 6 April 2012 3:51 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] issue with Digium TDM410P On Thu, Apr 05, 2012 at 08:14:09AM -0400, Mathieu Therrien wrote: Thanks for answer. Devices PAP2 and SPA3000 works good and use FSK for VMWI and CallerID, so I think TDM410P should also works. CallerID works good. So issue should be in source code of VMWI. Alec, If FSK is used for VMWI on his working setup, this doesn't sound like it would be fixed by your Generate VMWI neon pulses from FXS module.. patch to the wctdm24xxp DAHDI driver [1] I'm assuming. Must be something in chan_dahdi? But Now that the changes to the wctdm24xxp have calmed down (for awhile now... :/) it looks like it might be time to try again to rebase those patches on the current trunk. [1] https://reviewboard.asterisk.org/r/1144/ Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
