Re: [asterisk-users] Register = plain text password
Thanks A. J. *José Pablo Méndez * On Wed, Jan 22, 2014 at 3:22 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Wednesday 22 January 2014, José Pablo Méndez Soto wrote: Hello, Is there anyway to encrypt or scramble a bit the secret used to register with a provider? Im talking about the register = fromuser@fromdomain:secret@host directive in sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf No. Well. You *could* scramble it for storage; but that would only lull you into a false sense of security, because ultimately it would have to be able to be unscrambled by a program that was already right there on the machine, somewhere under /usr/src/ where any competent programmer can look at it. The client *has* to know the password in plaintext (or at least, how to decrypt the stored, encrypted password), in order to be able to send it to the server. The way things stand, the configuration file with the password in it need only be readable by the root user. And you know it has a password in it, so you take care with it. Here is an explanation from the developers of the Pidgin IM client, as to why they store passwords in plaintext in their configuration file: https://developer.pidgin.im/wiki/PlainTextPasswords This clever dude modified the code back in 1.4: http://www.oneharding.com/voip/asterisk_md5_register.html Unfortunately, that doesn't work. It just elevates a stolen hash to the same level of usefulness as a stolen password (and she even says so much, in the linked article). I imagine that so many years later, and now with the implementation of pjsip this secret could be better protected? No, because the underlying problem -- that decrypting a stored password also requires the decryption key; but if the decryption key and encrypted password are stored on the same machine, then anyone with access to the machine is able to decrypt the password -- is a limitation of the universe, *not* a limitation of present-day technology. There is simply nothing that anybody could invent that would get around this. It is very unsafe to keep the accounts password right out there. Any ideas? It's hidden behind another password, and that's about as secure as it's mathematically possible ever to make it. And if someone else has root access to your machine, then I humbly suggest that a SIP password might not be the driest lentil you have to soak. -- AJS Answers come *after* questions. -- _ -- 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
[asterisk-users] Register = plain text password
Hello, Is there anyway to encrypt or scramble a bit the secret used to register with a provider? Im talking about the register = fromuser@fromdomain:secret@host directive in sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf This clever dude modified the code back in 1.4: http://www.oneharding.com/voip/asterisk_md5_register.html I imagine that so many years later, and now with the implementation of pjsip this secret could be better protected? It is very unsafe to keep the accounts password right out there. Any ideas? *José Pablo Méndez * -- _ -- 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] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
Im using the one that comes with Ubuntu Server 10.10 (0.0.6~pre12-1): http://packages.ubuntu.com/search?keywords=libspandspsearchon=namessuite=mavericksection=all And having a sweet time with T.38 gateway. Oneiric already offers latest pre18. *José Pablo Méndez * On Wed, Jan 11, 2012 at 12:39 AM, Olivier oza_4...@yahoo.fr wrote: Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards -- _ -- 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] Problems faced in load testing of asterisk
I have given the rtp port range as 6000 to 8000 in rtp.conf. Is this not sufficient for running 1000 calls. Only even ports will be used for RTP I think, odd ports are reserved for RTCP, although I don't know how SIPp behaves in this line. 2000 ports should be reduced to 1000 ports following my theory. -- _ -- 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] Change port from 5060 on Snom phone
Can you email me off list (since this isn't really Asterisk related and a snom support issue, which I can help with) with some details and ideally a SIP trace? cheers, Paul. Closing this question with a final message including the [SOLVED] phrase will definitely help the community I think. At least I am interested in knowing the answer or final conclusion, would appreciated that very much. Regards, *José Pablo Méndez* -- _ -- 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] Change port from 5060 on Snom phone
Interestingly enough, they just list port 5060 in the Asterisk interoperability guide: http://wiki.snom.com/Interoperability/PBX/Asterisk Could that mean it is a fixed setting? (crappy) *José Pablo Méndez * On Fri, Jan 6, 2012 at 10:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm experimenting with using a port other than 5060 on one of our asterisk servers. Does anyone know how to change the target port on a Snom phone. I have tried adding :new port number to the end of the registrar but this doesn't work. Advanced - SIP/RTP - Network identity(port) is something else before anyone suggests it. Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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
[asterisk-users] Why write your dialplan using Lua?
Hello, Reading through the Wiki: Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk My question is, what is the benefit of using Lua? I recently noticed that OpenSIPS added a compatibility module to use Lua as well. However, where is the real advantage here? I mean, you have all these pieces in Asterisk like Lego blocks: AGI Commands, AMI Actions, Dialplan Applications, Dialplan Functions which I think are the ones that really limit what the PBX can do right? What's the difference between calling them out from extensions.conf or even from extensions.ael and calling them from extensions.lua? What else can you do writing you dialplan in Lua? Could I maybe program N-Way calling* with it? Are we talking about expanding Asterisk capabilities a huge deal? Or just performance wise during dialplan execution? With OpenSIPS I understand its power because you may affect the SIP behavior based on db queries performed by Lua scripts, or modify the next message to be sent, but with Asterisk, you wouldn't be able to modify an ongoing session through Lua based on, e.g., an incoming INVITE to establish a new conference room, would you? Or tell Asterisk to save the Register-CallID of an endpoint as part of the sip peer settings in memory, like the contact ID field. Regards, *José Pablo Méndez **By N-Way calling I mean having one party on hold, calling another party, and immediate (ad-hoc) invoke a conference bridge to join the 3 parties. Just like Cisco phone systems do. * * -- _ -- 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] Where are the fax instructions?
Kevin, I rolled back clean my virtual asterisk testbed and instead of libspandsp2 and libspandsp-dev, I installed libtiff4 - libtiff4-dev and downloaded/compiled spandsp 0.0.6, then I fixed the libraries with echo /usr/local/lib /etc/ld.so.conf.d/spandsp.conf cuz it wasn't able to load the module at all. Now it does but still can't see any reference to gateway support. Installed Asterisk 10 package, ./configure proves good spandsp support: checking for minimum version of SpanDSP... yes checking for span_set_message_handler in -lspandsp... yes checking spandsp.h usability... yes checking spandsp.h presence... yes checking for spandsp.h... yes checking for t38_terminal_init in -lspandsp... yes checking for spandsp.h... (cached) yes I decided to give 10.0 a try since res_fax_spandsp is included in the native code right? (if that even has any sense). It now shows: CLI fax show capabilities Registered FAX Technology Modules: Type: Spandsp Description : Spandsp FAX Driver Capabilities: SEND RECEIVE T.38 G.711 GATEWAY 1 registered modules I would consider this one solved but could you please throw some insight as to why 1.8.8.0 and 1.8.7.1 fail to build the gateway support? Thanks again, *José Pablo Méndez * On Thu, Jan 5, 2012 at 10:04 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/05/2012 01:03 AM, José Pablo Méndez Soto wrote: Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/**wiki/display/AST/T.38+Fax+**Gatewayhttps://wiki.asterisk.org/wiki/display/AST/T.38+Fax+GatewayI wrote this in my extensions.conf: exten = 306,1,NoOp(Fax transmission) same = n,Set(FAXOPT(gateway)=yes) same = n,Dial(DAHDI/3)-FXS port to fax machine same = n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -- Asterisk with TDM410 -- FXS -- Analog fax machine I am totally lost about the use of this new gateway module in the dialplan. I think it loads ok: CLI fax show capabilities Registered FAX Technology Modules: Type: Spandsp Description : Spandsp FAX Driver Capabilities: SEND RECEIVE T.38 G.711 Your res_fax_spandsp module was built without gateway support; what version of Asterisk are you using? -- _ -- 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] Where are the fax instructions?
Ah ok, I got the incredible idea to go look into the make menuselect for a res_fax_spandsp option after reading this: http://lists.digium.com/pipermail/asterisk-dev/2010-September/046344.html I found it in 1.8, now you say it doesn't come with gateway support. Thanks for the clarification Kevin, *José Pablo Méndez * I would consider this one solved but could you please throw some insight as to why 1.8.8.0 and 1.8.7.1 fail to build the gateway support? Asterisk 10 is the first version that includes T.38 gateway support. -- 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
Re: [asterisk-users] Why write your dialplan using Lua?
I wouldn't jump to a whole different language just to have an elegant script plan. There must be another reason why Lua is being so widely implemented than elegance and execution performance. Anyone? *José Pablo Méndez * On Fri, Jan 6, 2012 at 1:12 PM, Steve Edwards asterisk@sedwards.comwrote: On Fri, 6 Jan 2012, José Pablo Méndez Soto wrote: My question is, what is the benefit of using Lua? I've never used Lua, but I also have a curiosity about it. A couple of years ago, I wrote my first dialplan in AEL. Some bits were clumsy, minor syntax errors caused major parts of my dialplan to disappear, and I discovered a bunch of bugs*. But, many parts were elegant. Bits that would have been an ugly mess in 'plain dialplan' seemed obvious and clear. I would expect a similar experience with Lua. Kind of an opposite '1984**' experience. If you take the jump, please post your experiences. *) This was in 1.2 so there was no benefit in reporting the bugs. **) In Orwell's 1984, the 'scope' of language was reduced to eliminate the ability to express subversive thoughts. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Why write your dialplan using Lua?
Ok so its not a cosmetic thing only. I eases your administration. Do a point for performance. Now, what about my questions regarding extending the systems caps by building things asterisk could not build by itself. does it hold true? On Jan 6, 2012 3:28 PM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 6 Jan 2012, Danny Nicholas wrote: Not to purposely open a flame war here, but is lua preferable to ael? Aren't we better off just properly documenting our original conf files since new changes often introduce bad opportunities? One of the advantages of Lua is that it is embedded in other applications so you can 'leverage' the learning investment. Kind of like why you write your AGIs in Perl and I do in C. It's what we already know and we don't have to overcome 'yet another language' inertia. AEL was a big win for me over 'conf.' I'm hoping that Lua will be a big win over AEL. One thing does concern me... Here we are all discussing something we don't know anything about and nobody has chimed in with their experiences. If nobody is using it, how many 'arrows' will I get in my back when I blaze that trail? -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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-**usershttp://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] Where are the fax instructions?
Thanks from the bottom of my heart sir! Asterisk 1.8.7.1 is my build, I couldn't chose res_fax_spandsp from menuselect due to dependencies, so I installed libspandsp-dev: :~# apt-cache policy libspandsp* libspandsp2: Installed: 0.0.6~pre12-1 Candidate: 0.0.6~pre12-1 Version table: *** 0.0.6~pre12-1 0 libspandsp-dev: Installed: 0.0.6~pre12-1 Candidate: 0.0.6~pre12-1 Version table: *** 0.0.6~pre12-1 0 So, my dialplan snippet looks good, but I need to correct the lacking gateway capability? *José Pablo Méndez * On Thu, Jan 5, 2012 at 10:04 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/05/2012 01:03 AM, José Pablo Méndez Soto wrote: Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/**wiki/display/AST/T.38+Fax+**Gatewayhttps://wiki.asterisk.org/wiki/display/AST/T.38+Fax+GatewayI wrote this in my extensions.conf: exten = 306,1,NoOp(Fax transmission) same = n,Set(FAXOPT(gateway)=yes) same = n,Dial(DAHDI/3)-FXS port to fax machine same = n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -- Asterisk with TDM410 -- FXS -- Analog fax machine I am totally lost about the use of this new gateway module in the dialplan. I think it loads ok: CLI fax show capabilities Registered FAX Technology Modules: Type: Spandsp Description : Spandsp FAX Driver Capabilities: SEND RECEIVE T.38 G.711 Your res_fax_spandsp module was built without gateway support; what version of Asterisk are you using? 1 registered modules Also I have the FFA manual, which I couldn't understand. I think FAXOPT is common to both, but still not sure how to put them together. Where can I find documentation about configuring the call flow described? Fax for Asterisk is totally unrelated to T.38 gateway support. Or some insight will also be appreciated. Here is my sip peer config: [105](headquarters) ;zoiper phone type=friend secret= mailbox=105@default t38pt_udptl = yes Dahdi: ;FXS Modules group = 2 signalling = fxo_ks context = interno channel = 3-4 faxdetect = both Finally, a verbose output: == Using SIP RTP CoS mark 5 -- Executing [606@intern:1] NoOp(SIP/105-0002, Fax Transmission) in new stack -- Executing [606@intern:2] Set(SIP/105-0002, FAXOPT(gateway)=yes) in new stack [Jan 5 00:59:57] WARNING[1831]: res_fax.c:2783 acf_faxopt_write: channel 'SIP/605-0002' set FAXOPT(gateway) to 'yes' is unhandled! -- Executing [606@intern:3] Dial(SIP/605-0002, DAHDI/3) in new stack -- Called DAHDI/3 -- DAHDI/3-1 is ringing -- DAHDI/3-1 is ringing -- DAHDI/3-1 is ringing -- DAHDI/3-1 answered SIP/605-0002 -- Hanging up on 'DAHDI/3-1' -- Hungup 'DAHDI/3-1' == Spawn extension (intern, 606, 3) exited non-zero on 'SIP/105-0002' Since 'fax show capabilities' did not show that you have gateway support, you won't be able to enable it. That problem must be corrected first. -- 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-**usershttp://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] Best non polycom SIP conference room phone
Hello Bryant, Have you seen the snom meetingpoint? http://www.snom.com/en/products/sip-conference-phone/snom-meetingpoint/ I don't own one, but it looks like a fine piece of hardware. And snom is manufacturer of supported phones for Microsoft's Lync server (must say something their quality right?) http://technet.microsoft.com/en-us/lync/gg278172.aspx Doubles Polycom price though... *José Pablo Méndez * On Thu, Jan 5, 2012 at 11:19 AM, Bryant Zimmerman brya...@zktech.comwrote: I am looking for a really good SIP conference room phone for use with asterisk. I do not like Polycom at all. What would you all recommend? I have to be able to get them in the US. I found several that looked good but could not get them. And yes cost does matter but quality is the most important thing. Thanks Bryant -- _ -- 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
[asterisk-users] Where are the fax instructions?
Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten = 306,1,NoOp(Fax transmission) same = n,Set(FAXOPT(gateway)=yes) same = n,Dial(DAHDI/3)-FXS port to fax machine same = n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -- Asterisk with TDM410 -- FXS -- Analog fax machine I am totally lost about the use of this new gateway module in the dialplan. I think it loads ok: CLI fax show capabilities Registered FAX Technology Modules: Type: Spandsp Description : Spandsp FAX Driver Capabilities: SEND RECEIVE T.38 G.711 1 registered modules Also I have the FFA manual, which I couldn't understand. I think FAXOPT is common to both, but still not sure how to put them together. Where can I find documentation about configuring the call flow described? Or some insight will also be appreciated. Here is my sip peer config: [105](headquarters) ;zoiper phone type=friend secret= mailbox=105@default t38pt_udptl = yes Dahdi: ;FXS Modules group = 2 signalling = fxo_ks context = interno channel = 3-4 faxdetect = both Finally, a verbose output: == Using SIP RTP CoS mark 5 -- Executing [606@intern:1] NoOp(SIP/105-0002, Fax Transmission) in new stack -- Executing [606@intern:2] Set(SIP/105-0002, FAXOPT(gateway)=yes) in new stack [Jan 5 00:59:57] WARNING[1831]: res_fax.c:2783 acf_faxopt_write: channel 'SIP/605-0002' set FAXOPT(gateway) to 'yes' is unhandled! -- Executing [606@intern:3] Dial(SIP/605-0002, DAHDI/3) in new stack -- Called DAHDI/3 -- DAHDI/3-1 is ringing -- DAHDI/3-1 is ringing -- DAHDI/3-1 is ringing -- DAHDI/3-1 answered SIP/605-0002 -- Hanging up on 'DAHDI/3-1' -- Hungup 'DAHDI/3-1' == Spawn extension (intern, 606, 3) exited non-zero on 'SIP/105-0002' Thanks in advance for any help *José Pablo Méndez * -- _ -- 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 codec negotiation
Can you show us how the previous INVITE Looked like vs the current one? *José Pablo Méndez * On Sun, Jan 1, 2012 at 4:17 PM, cov...@ccs.covici.com wrote: Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message multiple audio streams not supported in the log. Is this by design? I found an issue 18859, but that referenced where the end point offered both regular rtp and srtp. But it seems to me if an endpoint offers various codecs, that asterisk could only complain if none of them match one that asterisk likes. If I only offer one codec, it works, but that seems an unnecessary restriction to me. Any assistance on this would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
Re: [asterisk-users] Use different local IP for each SIP trunk
May I ask why do you need different IP addresses to source calls? I mean, its not a common practice, would like to understand the idea behind it. *José Pablo Méndez * On Mon, Dec 19, 2011 at 11:07 PM, Anton Kvashenkin anton.juga...@gmail.comwrote: AFAIK you can add exterin= in sip.conf for each trunk, correct me if i'm wrong. 2011/12/20 Douglas Mortensen d...@impalanetworks.com Hello, ** ** I have a SIP provider whom I may want to have multiple trunks with, rather than just adding more channels to the individual trunk. I have discussed the matter with them they have told me that the only way that they identify which trunk should be used for each call is simply by the source IP address that the SIP calls are originating from. They do not use sip username/password or any other means to authenticate the remote caller. ** ** With that said, then it appears that the only way that I can have multiple trunks setup with them is to have asterisk use a different IP for all of the SIP RTP traffic for each given trunk. Essentially I would setup multiple IP addresses on my eth0 interface. Is there a way in asterisk that I could configure it to use one local IP for the source in all SIP/RTP traffic for 1 SIP trunk then a different local IP for the other SIP trunk? ** ** Thanks, - Doug Mortensen Network Consultant *Impala Networks Inc* CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ A.A.S. Information Technology . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 ** ** -- _ -- 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] Asterisk 1.8.7.2 now sends rport always
Thanks for answering Kevin. I guess my eyes were tired the night I started this thread, and yes, it would be ridiculous that Cisco phones couldn´t do rport. I actually found that its not the rport parameter, but the UDP ports usage. nat=no receives the REGISTER with source port 5400 for example, and the VIA header as 1.1.1.1:5060, so the 200 OK goes out with destination port 5060 as well, this works for the Cisco phone. nat=yes|force_rport sends the 200 OK out to destination port 5400 instead. I was aware of the change introduced because it was also mentioned in the asterisk-users] Asterisk 1.8.8.0 Now Available mail a few days ago, so I tried nat=no in the peer definition and it didn´t take effect for some damn reason and I got extra worried for nothing the past 24 hours to the point I couldn´t sleep (about to deploy 25 new phones with latest asterisk). Embarrassingly enough, I just tried the nat=no again both in the general and peer sections and the blessed phone registered My apologies, again, I wrote the thread late at night probably this blinded me. Now, one question about a previous answer from you (It is exactly that; 'force_rport' is now the default.): is the trigger for using the source UDP port from the REGISTER, inside the rport field and inside the destination UDP port of the 200 OK: 1. The mismatch between the UDP source port of the REGISTER and the VIA port? Or 2. The fact that the other entity sends an empty rport? 3. Or any of the above? Its a difficult question to ask/describe, so if I am not asking correctly please let me know. Thanks a lot, really. José Pablo Méndez On Sun, Dec 18, 2011 at 12:18 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/18/2011 01:42 AM, José Pablo Méndez Soto wrote: I have been testing with Cisco phones and have been able to register them with new firmware 9.2.1 (7911/7945/7970). All worked until I realized that from version 1.8.7.2, the VIA header contains the rport parameter, which breaks the phone registration process. Basically, the device can´t parse the VIA header this way, and when it gets the 200 OK to the REGISTER message containing the rport parameter, it refuses to process the registration internally, although it doesn´t complaint about it and Asterisk shows it as registered. First, let me say that it is pretty ridiculous that Cisco phones refuse to accept SIP responses with rport parameters in the Via header. But getting back to your problem... did you read the CHANGES file included with Asterisk 1.8.7.2? The *only* change between 1.8.7.1 and 1.8.7.2 was specifically handling of the 'nat' option in chan_sip to address a security vulnerability, but your message reads as if you are not aware of this. Asterisk 1.8.7.1 doesn´t behave this way and all works fine. The documentation about the use of the nat= parameter in sip.conf states: ;nat = no; Default. Use rport*if* the remote side says to use it. This is a bug in the sample configuration file; 'no' is no longer the default, 'force_rport' is. I understand that the other side must send an empty rport parameter to report the far end it needs the rport field to be filled in as per the RFC. The IP Phone is not sending the field at all, generating incongruity between the documentation and the real behavior. The only reason I think Asterisk would find the condition to be true, is due to a mismatch between the source port and VIA header ip:port inside the REGISTER message. Could this be the trigger of the 200 OK with rport (and, other SIP messages as well)? It is exactly that; 'force_rport' is now the default, and if you need 'no' behavior, you have to explicitly configure it that way. Can it be implemented a nat = never option in future releases? There is no need for such an option (which is why it was removed in the Asterisk 1.6.x timeframe). -- 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] Asterisk 1.8.7.2 now sends rport always
Thank you. *José Pablo Méndez * On Sun, Dec 18, 2011 at 8:23 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 12/18/2011 01:22 PM, José Pablo Méndez Soto wrote: Embarrassingly enough, I just tried the nat=no again both in the general and peer sections and the blessed phone registered My apologies, again, I wrote the thread late at night probably this blinded me. No problem, we've all done that :-) Now, one question about a previous answer from you (It is exactly that; 'force_rport' is now the default.): is the trigger for using the source UDP port from the REGISTER, inside the rport field and inside the destination UDP port of the 200 OK: 1. The mismatch between the UDP source port of the REGISTER and the VIA port? Or 2. The fact that the other entity sends an empty rport? 3. Or any of the above? Its a difficult question to ask/describe, so if I am not asking correctly please let me know. Thanks a lot, really. Not at all. The trigger for Asterisk to respond to the port that the request was sent from, instead of the port listed in the top-most Via header, is *exactly* 'force_rport'. This causes Asterisk to behave as if the 'rport' parameter was included in the top-most Via header, which would be an explicit request from the sending UA for Asterisk to respond to the sending port (and also report back what the sending port was, but that's not part of the problem here). So, if the sending UA include an empty 'rport' parameter in its top-most Via header, *or* if the Asterisk has been told to act as if one had been included even if it wasn't, then Asterisk will respond to the perceived sending port; otherwise, it will respond to the port listed in the top-most Via header. As far as we know from our research before making this change, the Cisco phones in question are the only ones that send their requests from one port and require the responses to go back to a different port. All other phones that we (and others) use with Asterisk use the same port for both, which makes them quite easy to use behind NAT devices. The Cisco phone models you are dealing with won't work behind a NAT device unless that NAT device has a 'helper' that understands SIP and can fix up this situation (and of course many Cisco phone users have Cisco routers that do exactly this). -- 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-**usershttp://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
[asterisk-users] Asterisk 1.8.7.2 now sends rport always
Hey, I have been testing with Cisco phones and have been able to register them with new firmware 9.2.1 (7911/7945/7970). All worked until I realized that from version 1.8.7.2, the VIA header contains the rport parameter, which breaks the phone registration process. Basically, the device can´t parse the VIA header this way, and when it gets the 200 OK to the REGISTER message containing the rport parameter, it refuses to process the registration internally, although it doesn´t complaint about it and Asterisk shows it as registered. Asterisk 1.8.7.1 doesn´t behave this way and all works fine. The documentation about the use of the nat= parameter in sip.conf states: ;nat = no; Default. Use rport* if* the remote side says to use it. I understand that the other side must send an empty rport parameter to report the far end it needs the rport field to be filled in as per the RFC. The IP Phone is not sending the field at all, generating incongruity between the documentation and the real behavior. The only reason I think Asterisk would find the condition to be true, is due to a mismatch between the source port and VIA header ip:port inside the REGISTER message. Could this be the trigger of the 200 OK with rport (and, other SIP messages as well)? Can it be implemented a nat = never option in future releases? I believe this is of utmost importance as many deployments are based on Cisco phones nowdays. Thanks. *José Pablo Méndez * -- _ -- 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] Chan_sip How to store Register Call ID?
Hello, I am trying to find a way to store the Register Call ID along with the peer info, or at least extract it from a log. What can be tweak in chan_sip to accomplish this? To illustrate, if the phone REGISTER message Call-ID header was something like 002584a2-58e40003-5b7b478e-f56e8005@192.168.1.200, then I would like to retrieve it somehow for that peer. I guess this might have some complications if multiple SIP clients register from the same IP, but the contact header can be used to differenciate. I see this is store along the Peers info and shows up from a sip show peer I checked thesecurity_events.h project, it seems like it is still disabled in Asterisk 1.8. Maybe if I could match REGISTER messages from that API, I could initiate a logging action somewhere. Any suggestion will be greatly appreciated *José Pablo Méndez * -- _ -- 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] How to install the new cdr-stats?
Hello, I wen't through a lot of pain as well. Please try this script if you can run your Asterisk installation on Ubuntu. The script is based on Areski's own script. Works flawlessly on server 10.10 and desktop 10.10 for me, but would like to fix any possible bugs when used on different platforms. Please comment if useful! *José Pablo Méndez * install-cdr-stats-ubuntu.sh Description: Bourne shell script -- _ -- 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] Has anybody been able to install CDR-Stats all the way through?
I have been trying to install cdr-stats for a week now, but there is no documentation worth the try and the amount of errors is huge. CUrrently stuck running python manage.py runserver 0.0.0.0:8000 I get python manage.py runserver 0.0.0.0:8000 Error: No module named dilla When starting apache, I get # /etc/init.d/apache2 start * Starting web server apache2 Syntax error on line 9 of /etc/apache2/sites-enabled/cdr_stats.conf: Invalid command 'PythonHandler', perhaps misspelled or defined by a module not included in the server configuration Action 'start' failed. The Apache error log may have more information.-- none! -- _ -- 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] Templates
Hi, Trying to create templates that allow higher compression of sip.conf, so for example: [internal-number](!) type=friend secret=bigsecret host=dynamic context=internal disallow=all allow=ulaw [100](internal-extensions) mailbox=100@internal-extensions [101](internal-extensions) mailbox=101@internal-extensions [102](internal-extensions) mailbox=102@internal-extensions The mailbox= parameter, as many others like username=, need a unique value. In my case, the sip profiles are very straight forward, I would like to know if I can use variables of some sort like this: [internal-extensions](!) mailbox=$[user]@internal-extensions [user](internal-extensions) -- automatically drags the mailbox= parameter filling it with the value between square brackets by virtue of the statement in the template Is this possible? *José Pablo Méndez * -- _ -- 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] Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory
Yes sir, We are pass the error. Works like a charm. I just documented this on our new wiki: http://voipcomsolutions.com/wiki/index.php?title=How_to_install_Asterisk_from_source_-_Google_Integration_ready Thanks again *José Pablo Méndez * 2010/12/1 José Pablo Méndez Soto aux...@gmail.com Thank you sir, I got to read your email a few minutes ago. I will try your recommendation and update. On Tue, Nov 30, 2010 at 5:01 PM, Tilghman Lesher tles...@digium.comwrote: On Tuesday 30 November 2010 16:27:33 José Pablo Méndez Soto wrote: Sorry never mind! I got it to work after sof-linking to /lib/, and loading res_jabber.so first, chan_gtalk.so second. So in summary: ln -s /usr/local/lib /lib/ The better way to do this would be: echo /usr/local/lib /etc/ld.so.conf.d/iksemel.conf ldconfig -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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
[asterisk-users] Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory
Hello, Can't get chan_gtalk.so module to load, neither res_jabber.so: Asterisk*CLI module load chan_gtalk.so Unable to load module chan_gtalk.so Command 'module load chan_gtalk.so ' failed. [Dec 1 16:10:05] WARNING[2931]: loader.c:387 load_dynamic_module: Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory [Dec 1 16:10:05] WARNING[2931]: loader.c:839 load_resource: Module 'chan_gtalk.so' could not be loaded. I got pass the module compilation after installing iksemel from tar ( http://code.google.com/p/iksemel/). Menuselect showed chan_gtalk check-able instead of XXX, which is good AFAIK. Also, Asterisk recognizes the modules just fine: Asterisk*CLI module load res_ res_adsi.sores_ael_share.so res_agi.so res_clialiases.so res_clioriginate.sores_convert.so res_crypto.so res_fax.so res_jabber.so res_limit.so res_monitor.so res_musiconhold.so res_mutestream.so res_phoneprov.so res_realtime.so res_rtp_asterisk.sores_rtp_multicast.so res_security_log.so res_smdi.sores_speech.so res_stun_monitor.so res_timing_dahdi.sores_timing_pthread.so res_timing_timerfd.so res_calendar.so Asterisk*CLI module load ch chan_agent.so chan_bridge.so chan_gtalk.so chan_iax2.so chan_jingle.so chan_local.so chan_mgcp.so chan_multicast_rtp.so chan_oss.so chan_phone.so chan_sip.sochan_skinny.so chan_unistim.sochan_dahdi.so Also, I made sure SSL libraries are in place: r...@asterisk:/etc/asterisk# dpkg -l openssl* libssl* ||/ NameVersion Description +++-===-===-== unlibssl none (no description available) ii libssl-dev 0.9.8g-16ubuntu3.4 SSL development libraries, header files and documentation ii libssl0.9.8 0.9.8g-16ubuntu3.4 SSL shared libraries unlibssl08-devnone (no description available) unlibssl09-devnone (no description available) unlibssl095a-dev none (no description available) unlibssl096-dev none (no description available) ii openssl 0.9.8g-16ubuntu3.4 Secure Socket Layer (SSL) binary and related cryptographic tools un openssl-doc none (no description available) iksemel was successfully installed: r...@asterisk:/etc/asterisk# ls /usr/local/lib/ libiksemel.a libiksemel.la libiksemel.so libiksemel.so.3 libiksemel.so.3.1.1 pkgconfig python2.6 Should I soft-link this libraries at another directory for Asterisk to find them? I found where chan_gtalk.so module gets the libraries from: r...@asterisk:/usr/lib/asterisk/modules# ldd chan_gtalk.so ldd chan_gtalk.so linux-vdso.so.1 = (0x7fff61bff000) libiksemel.so.3 = (Not found) libssl.so.0.9.8 = /lib/libssl.so.0.9.8 (0x7f7fd4ee6000) libcrypto.so.0.9.8 = /lib/libcrypto.so.0.9.8 (0x7f7fd4b5e000) libpthread.so.0 = /lib/libpthread.so.0 (0x7f7fd4942000) libc.so.6 = /lib/libc.so.6 (0x7f7fd45d2000) libdl.so.2 = /lib/libdl.so.2 (0x7f7fd43cd000) libz.so.1 = /lib/libz.so.1 (0x7f7fd41b6000) /lib64/ld-linux-x86-64.so.2 (0x7f7fd5559000) So I soft-linked under /lib/, and get a different error when loading the module: Asterisk*CLI module load chan_gtalk.so Unable to load module chan_gtalk.so Command 'module load chan_gtalk.so ' failed. [Dec 1 16:28:26] WARNING[3055]: loader.c:449 load_dynamic_module: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Dec 1 16:28:26] WARNING[3055]: loader.c:839 load_resource: Module 'chan_gtalk.so' could not be loaded. r...@asterisk:/usr/lib/asterisk/modules# !ldd ldd chan_gtalk.so linux-vdso.so.1 = (0x7fff61bff000) libiksemel.so.3 = /lib/libiksemel.so.3 (0x7f7fd5135000) --- It finds the library allright! libssl.so.0.9.8 = /lib/libssl.so.0.9.8 (0x7f7fd4ee6000) libcrypto.so.0.9.8 = /lib/libcrypto.so.0.9.8 (0x7f7fd4b5e000) libpthread.so.0 = /lib/libpthread.so.0 (0x7f7fd4942000) libc.so.6 = /lib/libc.so.6 (0x7f7fd45d2000) libdl.so.2 = /lib/libdl.so.2 (0x7f7fd43cd000) libz.so.1 = /lib/libz.so.1 (0x7f7fd41b6000) /lib64/ld-linux-x86-64.so.2 (0x7f7fd5559000) Any thoughts? *José Pablo Méndez * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
[asterisk-users] Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory
Sorry never mind! I got it to work after sof-linking to /lib/, and loading res_jabber.so first, chan_gtalk.so second. So in summary: ln -s /usr/local/lib /lib/ asterisk-climodules load res_jabber.so asterisk-climodules load chan_gtalk.so Cheers! *José Pablo Méndez * -- _ -- 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] Incoming calls through SS7 for data modem transmissions - possible??
Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk SS7 directly. We are planning on solving this by placing an Asterisk server with some TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the AS5300 for the dial-up to complete after authenticating against a RADIUS server. My questions is: can we use only Asterisk to complete/terminate the dial-up connection, removing the AS5300 out of the picture? Current topology to be set-up: Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet Ideal topology: Telco -- SS7 -- TE410P-AsteriskServer -- Internet Some posts talk about zapRAS being able to accomplish this, not quite sure though Sounds like possible: http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.htmlasterisk-users@lists.digium.com Sounds like not possible: http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html Thanks in advance, *José Pablo Méndez * -- _ -- 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] Incoming calls through SS7 for data modemtransmissions - possible??
Thanks Cary, What happens is, the Telco won't allow the small company to resell the ISDN connections, meaning, they bought the trunks and DIDs, then sold dialing plans to route incoming calls through the PRIs out the Internet. This is not the issue though. We definitely have to migrate to an SS7 capable platform, because that is the only way the Telco allows them to resell the dial-up connections (not ISDN), and Asterisk is the current bet. If we can get Asterisk to pick up those calls via SS7, then authenticate them, send them out to the Internet, we would be achieving a %100 usage on the Digium cards, because one of them wouldn't be used to talk to the AS. Can Asterisk do this? Thanks again, *José Pablo Méndez * On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch ca...@usawide.net wrote: I am not sure where you are and what legal conventions are involved. Are you saying the Telco (and legal restrictions) say you can’t send calls to the internet via the AS5300 but you can if Asterisk does it directly? What is the “logic” in that? Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? Or are you concerned about Asterisk handling the TDM to IP conversion in an adequate manner? I am not sure/aware myself that Asterisk will do a modem to IP conversion. I think in your example the AS5300 is doing that. What is the Telco’s problem in doing what the customer was doing before? Feel free to correspond directly if you want to. Cary Fitch -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez Soto *Sent:* Wednesday, November 24, 2010 7:31 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible?? Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk SS7 directly. We are planning on solving this by placing an Asterisk server with some TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the AS5300 for the dial-up to complete after authenticating against a RADIUS server. My questions is: can we use only Asterisk to complete/terminate the dial-up connection, removing the AS5300 out of the picture? Current topology to be set-up: Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet Ideal topology: Telco -- SS7 -- TE410P-AsteriskServer -- Internet Some posts talk about zapRAS being able to accomplish this, not quite sure though Sounds like possible: http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.htmlasterisk-users@lists.digium.com Sounds like not possible: http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html Thanks in advance, *José Pablo Méndez** * -- _ -- 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] Incoming calls through SS7 for datamodemtransmissions - possible??
Thanks Cary, The first topology we are working on should be the best way then. Asterisk will answer SS7 calls, route them to the ISDN channels to be terminated by the AS5300 as they were doing before. I think TDM-2-TDM shouldn't be that much of a problem eh? No further equipment needed? *José Pablo Méndez * 2010/11/24 Cary Fitch ca...@usawide.net I understand the problem. You can’t resell PRI connections. I don’t think Asterisk can convert TDM to IP. It does convert TDM to SIP which is then sent out over IP.What you want to do is have it do the TDM/Modem conversion without the PRIs and Cisco Gear. There used to be a way to do this, and maybe still is but not just with Asterisk perhaps. I know that Ascend/Lucent TNTs (and I am sure some other equipment) could take TDM trunks, which could be SS7 trunks, and convert them to IP. The point in this is that they are SS7 based. You can take SS7 trunks from either the Asterisk box or direct from the Telco and convert them to IP. NO PRIs involved. Yes, more “telco grade carrier equipment” but no PRIs. A lot of this equipment was available by the pound a few years back. Cary -- *From:* José Pablo Méndez Soto [mailto:aux...@gmail.com] *Sent:* Wednesday, November 24, 2010 8:34 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* ca...@usawide.net *Subject:* Re: [asterisk-users] Incoming calls through SS7 for datamodemtransmissions - possible?? Thanks Cary, What happens is, the Telco won't allow the small company to resell the ISDN connections, meaning, they bought the trunks and DIDs, then sold dialing plans to route incoming calls through the PRIs out the Internet. This is not the issue though. We definitely have to migrate to an SS7 capable platform, because that is the only way the Telco allows them to resell the dial-up connections (not ISDN), and Asterisk is the current bet. If we can get Asterisk to pick up those calls via SS7, then authenticate them, send them out to the Internet, we would be achieving a %100 usage on the Digium cards, because one of them wouldn't be used to talk to the AS. Can Asterisk do this? Thanks again, *José Pablo Méndez** * On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch ca...@usawide.net wrote: I am not sure where you are and what legal conventions are involved. Are you saying the Telco (and legal restrictions) say you can’t send calls to the internet via the AS5300 but you can if Asterisk does it directly? What is the “logic” in that? Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? Or are you concerned about Asterisk handling the TDM to IP conversion in an adequate manner? I am not sure/aware myself that Asterisk will do a modem to IP conversion. I think in your example the AS5300 is doing that. What is the Telco’s problem in doing what the customer was doing before? Feel free to correspond directly if you want to. Cary Fitch -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez Soto *Sent:* Wednesday, November 24, 2010 7:31 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible?? Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk SS7 directly. We are planning on solving this by placing an Asterisk server with some TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the AS5300 for the dial-up to complete after authenticating against a RADIUS server. My questions is: can we use only Asterisk to complete/terminate the dial-up connection, removing the AS5300 out of the picture? Current topology to be set-up: Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet Ideal topology: Telco -- SS7 -- TE410P-AsteriskServer -- Internet Some posts talk about zapRAS being able to accomplish this, not quite sure though Sounds like possible: http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.htmlasterisk-users@lists.digium.com Sounds like not possible: http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html Thanks in advance, *José Pablo Méndez** * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] [asterisk-ss7] Incoming calls through SS7 for data modem transmissions - possible??
Thank you Horacio and Cary. We will try receiving SS7, routing via SIP, answering on the AS5300, then looping back to itself (out PRI, in PRI ports) in order to invoke the modem termination. This way we may be able to spare the TDM cards in Asterisk and reuse the E1 ports installed in the gateway. Best regards, *José Pablo Méndez * 2010/11/24 Horacio J. Peña hor...@compendium.com.ar Hola! ZapRAS seems to work only with ISDN calls. This command is not for use with analog lines; it does not provide a modem emulator. (http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS) You need something doing the modulation. It seems that iaxmodem is your best bet, and you'll have to make a good bunch of work on it to be able to use as you want to. If your client has the cisco gateways, I'd suggest you to keep them. They are very reliable and tested, and with MICA cards they have not a high resale value, so you'll probably end with them as paperweights unless you happen to have some stack of C549 cards to repurpose them. Saludos, H On Wed, Nov 24, 2010 at 07:58:37PM -0600, José Pablo Méndez Soto wrote: Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks from, discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk SS7 directly. We are planning on solving this by placing an Asterisk server with some TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the AS5300 for the dial-up to complete after authenticating against a RADIUS server. My questions is: can we use only Asterisk to complete/terminate the dial-up connection, removing the AS5300 out of the picture? We would probably need a PPP channel configuration to link the modem connection with the Internet. Current topology to be set-up: Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet Ideal topology: Telco -- SS7 -- TE410P-AsteriskServer -- Internet Some posts talk about zapRAS being able to accomplish this, not quite sure though Sounds like possible: [1] http://lists.digium.com/pipermail/asterisk-users/2004-January/026956 .html [2] http://lists.digium.com/pipermail/asterisk-users/2009-November/24021 8.html Sounds like not possible: [3] http://lists.digium.com/pipermail/asterisk-users/2009-November/24020 2.html Thanks in advance, José Pablo Méndez References 1. mailto:asterisk-users@lists.digium.com 2. http://lists.digium.com/pipermail/asterisk-users/2009-November/240218.html 3. http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 -- Horacio J. Peña hor...@compendium.com.ar hor...@uninet.edu -- _ -- 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
