Re: [asterisk-users] Movistar sip Mexico
It is possible that Asterisk requires an rtpmap even for static payload types (I'm not sure about this). The INVITE from your provider omits rtpmap for payload type 18 (G729) which is perfectly valid. On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez dgonza...@denwaip.comwrote: Hello, Thanks for the quickly response. I have only G729 in the peer but I have t38pt_udptl= yes If I put t38pt_udptl=no , asterisk reject the call with 488 code. The problem is that Movistar send T38 codec in all calls and I need ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have only T38 I have to negociate a fax call. Thanks. On Wed, Nov 20, 2013 at 4:46 PM, Alyed al...@vivoxie.com wrote: Think you only need to make sure you have in your sip.conf file these configs: [your-device-name] . . disallow=all allow=g729 . . Alyed 2013/11/20 Damian Gonzalez dgonza...@denwaip.com Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 6372 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy How can I ignore T38 and use only G729 for this call?. Thanks for your help. Damian -- -- _ -- 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 -- Kristian Kielhofner -- _ -- 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] Somewhat-OT: Stupid NAT tricks to learn from Apple?
I've been spending some time looking at some of the significant changes Apple has made to Facetime in iOS 7. I'm far from an Apple fanboy but some of them are pretty interesting: - multiplexing everything over a single UDP port - deflate compression with SIP - various /slight/ protocol violations ;) More here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html As SDP bodies swell more and more can we hope to build significant support for multiplexing and deflate compression in the SIP-focused open source ecosystem? -- Kristian Kielhofner -- _ -- 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] Somewhat OT: Specific SIP packets can cause ethernet controller reset
Update with a response to the statement from Intel: http://blog.krisk.org/2013/02/packets-of-death-update.html On Wed, Feb 6, 2013 at 11:08 AM, Kristian Kielhofner k...@kriskinc.com wrote: While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner -- Kristian Kielhofner -- _ -- 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] Somewhat OT: Specific SIP packets can cause ethernet controller reset
While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner -- _ -- 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] Wireshark AMI Dissector
Hello everyone, Does anyone know of a Wireshark AMI (Asterisk Manager Interface) dissector? Decode as telnet and display filter telnet.data kind of work but TCP reassembly can't happen without a better understanding of the protocol... Thanks! -- Kristian Kielhofner -- _ -- 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] directrtp with SIP + H.323
On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis supp...@ocg.ca wrote: We're creating a SIP gateway for a client that will take one leg of a call in via SIP, and out the other side via H.323. To minimize load on the gateway, we would like to have the RTP stream bypass the gatewayy altogether (directrtp/reinvite). Is this possible with these to protocols? Thanks Yate claims it can do this: http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] large scale paging
On Fri, Feb 5, 2010 at 4:50 PM, Mark Willis marksli...@markwillis.net wrote: Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097 What you really want is multicast RTP, preferably as implemented in SNOM phones: https://issues.asterisk.org/view.php?id=11797 http://wiki.snom.com/Settings/multicast_address One RTP stream, any number of receivers, no SIP session. Doing this with unicast RTP and individual INVITEs would be tough. If your system can't do 500 call setups per second (or better) you'll introduce massive delays in call setup to the recipients, not to mention serious RTP burden with that many streams. I hope you haven't bought phones yet (or bought Snom) ;). -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of 603 Declined
On Sat, Jan 30, 2010 at 9:03 AM, Kevin P. Fleming kpflem...@digium.com wrote: Olle E. Johansson wrote: Here's something interesting: 21.4.25 487 Request Terminated The request was terminated by a BYE or CANCEL request. This response is never returned for a CANCEL request itself. This is only used in combination with Cancel's, but in this case the call was cancelled by the dialplan, not by the caller. It's a misuse, but a bit clever one. Yes, I think 487 seems to be a logical choice here; it's very close to what 487 is normatively used for. Now, I realize that we also need a setting to indicate whether Asterisk is authorative for a domain or not. If we're the only owners of a domain, we should generate 6xx class errors, if not, 4xx error. So this also applies to 486/600 busy 488/606 and 404/604. If we start separating 4xx and 6xx replies, we might as well do it right. So the domain configuration needs an option per domain whether we're just part of a cluster handling a domain or if we're THE domain handler. That would be a good idea, yes. The quick fix for my issue is to make Asterisk send 487. The real fix, however, looks to be much broader (configuration option for domains, etc). How much work is this? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of 603 Declined
On Fri, Jan 29, 2010 at 5:59 AM, Olle E. Johansson o...@edvina.net wrote: 29 jan 2010 kl. 10.25 skrev Alex Balashov: I don't know about 4xx, but 503 would be more benign for general/ miscellaneous errors than 603. 503 indicates that there's a problem with the server, so that's not a good replacement. We're sending this when there's a failed call, in most cases because of the outbound channel failure without a proper hangup cause being set in that channel driver (may very well be chan_sip :-)) No, we need something in the 4xx class. I haven't had time to go through and consider all the options in the massive set of RFCs we have to work with, but will try to do that tonight after a Friday night dinner - if no one on the list comes up with the solution before that. Isn't that a hacker way of spending Friday night - enjoying the wonderful prose of RFC3261 and companions? /O Olle, I don't want to ruin your plans for tonight (RFC3261 is a lot of fun) but how about 403: 21.4.4 403 Forbidden The server understood the request, but is refusing to fulfill it. Authorization will not help, and the request SHOULD NOT be repeated. I like this because the most reliable way to get Asterisk to send a 603 at the moment is with something like this: sip.conf: [general] context=nocrackers extensions.conf: [nocrackers] exten = i,1,Hangup exten = s,1,Hangup exten = t,1,Hangup Wouldn't a 403 be perfect in this scenario? It looks like there are certainly other cases where it wouldn't fit quite as well (I haven't even looked at those involving REFER) but it looks perfect to me. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of 603 Declined
On Fri, Jan 29, 2010 at 10:31 AM, Kevin P. Fleming kpflem...@digium.com wrote: Well, that's the problem, and it's the reason why 603 is so commonly used. This is a situation where the current request has failed, but there is no indication that repeating the request will also fail. 403 means that the request should not be repeated without either changing it or authenticating as a different entity, which is a different scenario. This is true... Authenticating as a different entity would/could potentially match other peers (causing a 407) and probably isn't technically correct. However, if they didn't match an existing peer (to be challenged or not) using Asterisk's standard peer matching, how did they end up in the nocrackers context anyway? Either way I wasn't considering 5xx responses because of Olle's request. It is very likely that there is no standard-defined 4xx code for 'cannot process this call right now', only the 5xx and 6xx variants. Asterisk has certainly bent standards (which real world implementation hasn't) before. It seems to me that the best reply is the one that's most likely to encourage correct behavior from the far end... 603 almost certainly doesn't do that. In this scenario any forking proxy faced with a 603 coming from Asterisk has to break RFC compliance just to successfully complete the request on another host. Nasty. Are we back to the next-most-generic SIP error, 503 (as originally suggested by Alex)? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use of 603 Declined
Hello everyone,
I've had the time to examine some specific serial/parallel forking
scenarios with Asterisk lately. Looking at chan_sip it appears that
anytime Asterisk wants to tear down a call before it's brought up, it
sends a 603 Declined:
} else {/* Incoming call, not up */
const char *res;
if (p-hangupcause (res =
hangup_cause2sip(p-hangupcause)))
transmit_response_reliable(p,
res, p-initreq);
else
transmit_response_reliable(p,
603 Declined, p-initreq);
p-invitestate = INV_TERMINATED;
Obviously this doesn't include cases where the URI is not found, the
codec is incompatible, etc. More just general failure stuff like
executing Hangup() on an unanswered channel.
However, 6xxx responses are somewhat religious/political in the SIP
sphere... Being that they are global responses, how could this
single Asterisk instance know that this call is unacceptable
everywhere/anywhere? From RFC3261:
21.6.2 603 Decline
The callee's machine was successfully contacted but the user
explicitly does not wish to or cannot participate. The response MAY
indicate a better time to call in the Retry-After header field. This
status response is returned only if the client knows that no other
end point will answer the request.
I suppose manually executing Hangup() justifies the first statement
but it's the last sentence that bothers me:
returned only if the client (Asterisk) knows that no other end point
will answer the request
That's a little presumptive of the Asterisk system, don't you think?
;) While I don't have any better alternative responses I'm just
bothered by the global nature of 6xx failures in the first place.
Any thoughts?
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of 603 Declined
On Thu, Jan 28, 2010 at 4:23 PM, Alex Balashov abalas...@evaristesys.com wrote: It's also problematic because a 3261-compliant SIP proxy or UAC is not going to attempt to reach the destination by alternate means (serial forking in the case of the proxy, or a new call leg in the case of the UA) because of this precise implication of 6xx-class final replies. -- Alex This is precisely why some proxies (including OpenSIPS Kamailio) have added the disable_6xx_block parameter to specifically break this 3261-compliant behavior. Of course this being a global proxy parameter prevents cases where you really do want a 603 to stop forking. I've read that OpenSIPS is going to make it possible to activate this behavior via flags or some other means but in the meantime I'd like to see Asterisk be a little more flexible and um, friendly in this case. Luckily Asterisk is open source and I can make that change if I like but... A quick poll: Who thinks Asterisk should severely limit the cases it sends 6xx responses? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of 603 Declined
On Thu, Jan 28, 2010 at 4:52 PM, Alex Balashov abalas...@evaristesys.com wrote: I was just about to mention the disable_6xx_block parameter, but figured it would be too pedantic/off-topic for this thread. I didn't. Google has a great memory and hopefully now when some poor soul is researching this (Asterisk + OpenSIPS/Kamailio 6xx replies) they will find this thread to tie their solution together. I can't think of any cases where it should be used except where some sort of formal error arises, to be honest. When is Asterisk ever in an authoritative position to deem a destination certifiably unreachable except, perhaps, an invalid IP address, unresolvable host, or something of that sort? Agreed. Even then, on an incoming request, how would it know? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, Jan 13, 2010 at 2:43 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Thanks for that. Looking at the RTP packets I can see two types as you point out. The first appears to be the audio: Real-Time Transport Protocol 10.. = Version: RFC 1889 Version (2) Payload type: ITU-T G.711 PCMU (0) And as you say, the DTMF events are clear to see: RFC 2833 RTP Event Event ID: DTMF One 1 (1) ..00 1010 = Volume: 10 So, as these can be seen in the stream, do I need to tell Asterisk to detect these? Does it not do that when I set: dtmfmode=rfc2833 ??? There are some pretty widely recognized RFC2833 compatibility issues in the SIP/RTP world. Which version of Asterisk are you using? Do you know what kind of equipment your carrier is using? If they are using Asterisk you can try to add rfc2833compensate=yes to their peer entry in sip.conf. The SIP debug, however, will tell you if the remote end is configured to use RFC2833 or not. That's why I was telling you to look for telephone-event in the INVITE from your provider. Keep in mind SIP (most likely) runs over UDP between you and your provider, not TCP. I see a 'telephone-event' : a=rtpmap:101 telephone-event/8000 That's all you need to know. They are configured for RFC2833 and they're sending RFC2833. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: ..snip.. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] Just because they don't offer assistance with Asterisk doesn't mean they don't use it themselves. If you send me a packet capture in PCAP format with SIP+RTP between your system and your carrier I can debug this further. I appreciate this is a 'how long is a piece of string question Kristian, but is there likely to be a way I can fix this? You can try the rfc2833compensate option... Other than that I can't know until I see a packet capture. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, Jan 13, 2010 at 12:07 PM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: That may contain sensitive data, such as SIP account/password details - so I'll pass on that, but thanks for the offer. Even if they are using auth it's challenge response and fairly difficult to reverse engineer, not that I have the time for that. I do however, specialize in debugging DTMF. I always make time for interesting cases. I also own a voice service provider so it's unlikely I'm interested in your sipgate credentials :). Good luck. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net wrote: Codec? I've had 2833 do funny things with anything other than a/ulaw (might just be me though..) S -- Codecs other than G711u/a don't support inband DTMF. Seeing as INFO is rarely used that pretty much leaves RFC2833. Turn on SIP debugging and look in the INVITE from the provider for telephone-event. If you see it, they're configured to use RFC2833. If they are, you need to do a packet capture or other RTP debug to see the out of band RFC2833 events. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
dump shows nothing. SIPGATE have said;
you should be able to set the dtmfmode to rfc2833 in your default
sip.conf.
Best regards,
Frederik
I've tried other combinations such as info, inband et al. I'm guessing
{that's all it is} that rfc2833 will signal the dtfm over sip as opposed
to in the audio stream?
RFC2833 is carried in RTP like the audio stream. However, it uses a
different payload type from the RTP packets used to transport the
audio. If you did an RTP capture you would be able to see the RFC2833
events (which correspond to DTMF presses).
The SIP debug, however, will tell you if the remote end is configured
to use RFC2833 or not. That's why I was telling you to look for
telephone-event in the INVITE from your provider. Keep in mind SIP
(most likely) runs over UDP between you and your provider, not TCP.
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Semi-OT: Configuring SIP trunks with Cisco UCM 7.0.
Hello everyone, I'm trying to turn up a SIP trunk with a Cisco UCM (Unified Communications Manager/Call Manager). It's currently configured for 3rd party call control (3pcc). The INVITEs show up without an SDP... Neither the Cisco admin nor myself can find any documentation on how to disable this feature (3pcc). Does anyone happen to know how to disable 3pcc on Cisco Unified Communications Manager 7.0? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay in processing dtmf
On Mon, Oct 19, 2009 at 2:16 PM, Danny Nicholas da...@debsinc.com wrote: You might want to play with tonedur in dahdi.conf. This IME effects SIP calls as well. Configuration changes in dahdi.conf do not affect SIP channels. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Duplicate DTMF
On Wed, Sep 9, 2009 at 10:22 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I've come across a nasty problem just as we are ready to put our first system into production. During our final testing, we were plagued with several invalid extension or password incorrect messages when we knew the information entered was correct. Upon investigation, we saw that DTMF signals were occasionally but not consistently duplicated. We might dial extension 1234, see 1234 on the phone from which we dialed, but see 112334 on the Asterisk console. We have seen this from cell phones calling via the PSTN (we use a SIP trunking carrier and do not handle the PSTN interface ourselves); we've seen it from land line phones via the PSTN, and have even seen it internally from our own Snom SIP phones. dtmfmode=auto but we have also tried setting it to rfc2833 and we have tried relaxdtmf set to both yes and no. We are running Asterisk 1.6.1.6 on CentOS 5.3. We really don't know what more to do to fix it. Googling shows that others have had this problem but I haven't seen a clear resolution other than playing with relaxdtmf. How do we solve this problem? Thanks - John Fairly typical for most SIP carriers... My blog entry may be able to illuminate this a bit: http://blog.krisk.org/2009/02/update-youve-been-waiting-for.html In short RFC2833 DTMF issues are fairly common. It's troublesome enough when trying to go directly to the Tier 1 carriers themselves. More than likely you're dealing with a reseller (carrier) that most likely inherits issues from their carrier and adds their own. A couple of weeks ago someone e-mailed me asking for RFC2833 assistance with Asterisk and a carrier using Sonus. Turns out: a) The carrier was a reseller of various other carriers that use Sonus. b) The carrier proxied RTP (and therefor RFC2833 events) through an Asterisk 1.2 machine; further mangling the RFC2833 events. Other than some drastic changes at the carrier there wasn't much that could be done... Sorry I can't offer more specific advice to your situation bit without an RTP packet capture there isn't much I (we) can do. P.S. - Ignore any suggestions for gain, etc. These are for Zap channels and do not apply to sip. Changing anything in zapata.conf will not affect your situation. I'm not even sure of the existence or purpose of relaxdtmf in sip.conf in Asterisk 1.4 or later. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP reply CALL-ID from ITSP has internal address in host part
On Thu, Sep 10, 2009 at 12:51 PM, Olle E. Johansson o...@edvina.net wrote: You were lucky that you could disable it. I've met cases where firewalls have a smart SIP something feature that can't be disabled and enabled it was successful in disabling all media stream by publishing RFC1918 addresses on the outside. Ouch. /O +1 for SIP TLS/SIPS. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VUC: RE: Friday 11th: Aswath Rao: Trapezoidal VoIP is Evil on VoIP Users Conference at Noon EDT
On Thu, Sep 10, 2009 at 2:24 PM, Dean Collins d...@cognation.net wrote: It might just be me but what is trapezoidal voip? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). I don't know if I'll be able to make the call but my guess is he's referring to the SIP trapezoid: http://www.iptel.org/sip_trapezoid The SIP trapezoid is a concept/teaching tool usually used in situations involving a proxy (or multiple proxies) illustrating the SIP concept of distinct separation between signaling and media. Granted this isn't unique to SIP... ISDN, SS7, H.323, MGCP, and I'm sure others also make this distinction. IAX is the only protocol I know of that doesn't (which is where most of it's NAT advantages come from). He's probably going to talk about the advantages and disadvantages of the trapezoid although from the title I'm guessing he's going to focus on the disadvantages ;). Then again I could be completely wrong. The SIP trapezoid is real but this speculation is purely my own. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
On Mon, Apr 13, 2009 at 5:32 PM, John covici cov...@ccs.covici.com wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance 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 John, Assuming this is Asterisk 1.4 or later... The duration used by Asterisk is the same duration sent from the phone. The duration of those DTMF key presses should match the time the user is holding down the key. What type(s) of phones are these? You should also look into using Asterisk 1.4.24.1 or later (if you aren't already). There have been many improvements to the RTP code to better handle quirks with the equipment (especially Sonus) used by various providers. Assuming your provider is to spec (and so is your phone) your provider should not be complaining that the duration of your DTMF key presses are too short... With that being said AFAIK there is no way to specify a minimum duration for an RFC 2833 DTMF in Asterisk on a bridged channel. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SheevaPlug Development Kit
Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Of course one of my first tasks will be to get Asterisk running on it... ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM codec is a good choice ???
On Mon, Feb 23, 2009 at 6:59 PM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the pass-throu calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729 license, so I'm thinking to buy new IP phones with GSM support, so I have no problem with the voicemail system. Are the IP phone with GSM support a good choice for me ??? (Maybe in the future I need to connect the Asterisk with the PSTN, GSM doesn't matter at this point ???) Really thanks, Alejandro Install the G.729 sound files and make app_voicemail record messages (format=g729) in G729. As long as you don't need meetme or a few other apps that essentially require G.729 transcoding you don't need a license. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus
On Wed, Feb 4, 2009 at 4:00 PM, Gregory Boehnlein da...@nacs.net wrote: Hello, Is anyone running Asterisk 1.4 w/ RFC2833 to Level3's SONUS network? We are unable to get reliable RFC 2833 DTMF working, and have instead had to use G711ULAW w/ INBAND DTMF to get around the issue. Looks like an issue on the SONUS side. Anyone else have this issue? Welcome to the club! ;) I'll be blogging about this later today. Look out for that post... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 conection to a Cisco2600
On 1/28/09, Rafael RGV rafael.ri...@gmail.com wrote: Hi I am trying to connect asterisk with a Cisco GW 2600 with E1 pri using a Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02), Errors: [Jan 28 17:32:33] VERBOSE[6182] logger.c: == Primary D-Channel on span 1 up [Jan 28 17:32:33] WARNING[6182] chan_dahdi.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. Configure Asterisk for pri_net. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailinglists
On 1/27/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: I would have said Queen's English, but that evokes Freddy Mercury. ...and Freddy Mercury evokes Kevin Fleming. Perfect - we're back on topic! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packet8 hacked
On Fri, Jan 23, 2009 at 9:18 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Fine using ATT's (at least I think they belong to ATT) DNS servers (Also do NTP). I will make this thread useful to someone. Listed below because they are easy to remember and have never failed me even when a customer's ISP's DNS is down, I get the call Our Internet is Down but they can ping my servers by IP. Also, can speed up complaints of a slow network (if DNS lookups are the reason for the slowness. 4.2.2.1 4.2.2.2 4.2.2.3 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) Steve, Those are Level(3)s anycast DNS servers - part of internet legend! I once talked to a Level(3) engineer about how many queries those things do (combined) and it was amazing. I don't remember the exact number but it was something in the TENS OF MILLIONS/day, maybe more. It was one of those numbers (especially for daily traffic) that you just don't really comprehend. I've read (on NANOG, of course) that Level(3) actually encourages (at least doesn't discourage) their public use. They log the queries and sell the results for data mining purposes (DNS Alexa perhaps). -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] random Linksys question
On Thu, Jan 22, 2009 at 3:11 PM, Jeff LaCoursiere j...@jeff.net wrote: Can you configure the LAN port on the back of a 2102 to be bridged rather than routed to the WAN port? To my knowledge this is available on all Linksys ATA type devices that offer both ports. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 404 not found from one ip-adress
On Tue, Jan 13, 2009 at 9:59 AM, Ralf Träskman r...@adlibris.com wrote: Hi The provider dont use register, they are running openSER I have this in my sip.conf [outgoing] context=ip-only disallow=all allow=alaw,ulaw canreinvite=yes dtmfmode=rfc2833 host=sip.hub.ip-only.se insecure=very reinvite=yes type=friend [incoming] disallow=all allow=alaw,ulaw context=ip-only type=user Regards /ralf Ralf, That incoming peer isn't matching anything. They're probably hitting the context defined in [general]. Add another peer/friend match with the other servers IP/hostname and the ip-only context. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Friday VUC 12 Noon ET with Kristian Kielhofner: Identifying Asterisk Quality Issues
On 12/31/08, Trixter aka Bret McDanel trix...@0xdecafbad.com wrote: On Wed, 2008-12-31 at 18:39 -0500, C. Savinovich wrote: Hello Kris, what's up? I was reading your description for Recqual, and there is one thing I don't get: - We perform the test, which consists in playing a sound file over the network via asterisk, and then comparing the difference in quality. - How is it going to tell me what in my network is causing the loss of quality? indeed, you would want to differentiate between jitter and loss, as well as transcoding issues that may be present. Even that wont really tell you what part of the network, but it can tell you what to look for to try to figure it out. -- Trixter http://www.0xdecafbad.com Bret McDanel pgp key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x8AE5C721 (moved to Asterisk-Users) We spent quite a bit of time discussing recqual earlier today on The VoIP User's Conference. Thanks again Randy for organizing such a lively discussion every week! I had quite a bit of time to explain recqual, the environment it came out of, and where it could use some improvement. If you have the time I highly recommend at least /skimming/ the recording of the call. It will most likely answer any of your questions. To answer your specific questions: Recqual cannot possibly determine specific points of audio degradation in a complete audio path. It can only test one variable (basically). It is important that you use other, existing tools to try to eliminate all other potential sources of audio degradation. Eliminate your machine, local LAN, and other factors as much as possible. You will need to have at least two separate paths to the network you are testing (usually the PSTN), one of which is of known quality. Once all of these things are established, you can use recqual to test the quality of various other audio paths, regardless of technology (thanks to Asterisk). SIP providers, POTS lines, B channels on a PRI, etc can all be compared against your reference, whatever that might be. Recqual, up to this point, has only been used by the author within SIP networks. However, as I pointed out in the README, VUC, and earlier in this message, thanks to the protocol independent nature of Asterisk, virtually any technology can be tested with only a few modifications to your (possibly existing) Asterisk configuration. Also - recqual does not use sound files, as least not yet. That would be interesting although I don't know how it could work. As of now recqual simply generates continuous tones and does some comparisons on them after they have gone through the network being tested (again, usually the PSTN). I'm glad we have tools like Asterisk and ecasound that are simple enough for someone like me to slap a shell script on top of to do some pretty cool stuff! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday VUC 12 Noon ET with Kristian Kielhofner: Identifying Asterisk Quality Issues
On 12/31/08, randulo spamsucks2...@gmail.com wrote: Happy New Year in advance by a few ticks for the northern hemisphere. Here's the first topic and guest for 2009: In any voice path there are several potential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of the problem. Recqual is a utility that uses Asterisk and Ecasound to make comparisons of complete audio paths - whether they be IP, TDM, or any combination. In this Friday's VoIP User's Conference, Kristian Kielhofner will give an introduction to Recqual and provide several examples of its usage in real world contexts. More on Recqual here on Kristian's blog : http://tr.im/recqual2 or on Voip-Info: http://tr.im/recqual Join us if your head isn't still throbbing. Best to all of you out there, /r I'd like to add that I am recruiting devs and other experts to participate in this call. John Todd will be joining me along with several others (hopefully) to provide real world input on overall call quality topics, such as: - RTCP/RTCP XR - QoS - Etc, etc It's shaping up (no pun intended) to be a very interesting call! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why does users.conf generate SIP peer and SIP user?
On Tue, Dec 23, 2008 at 4:40 AM, Steve Totaro stot...@first-notification.com wrote: It's all ball bearings these days What is the deal with Fletch quotes these days? Don't get me wrong, I appreciate them but I'm starting to wonder where this is all coming from. I *think* it's because Fletch has been on HBO lately. Am I correct? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Tue, Dec 23, 2008 at 6:31 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote: Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: You seem to have ommited the relevant links: http://blog.krisk.org/2008/12/introducing-recqual.html http://admin.star2star.com/recqual/ Looks interesting... -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir Tzafrir, I included a link to my blog in the original post but it probably wasn't as clear as it could have been. Looks interesting? Thanks! I'd love to see it deployed more, I've only used it in a few specific scenarios. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Tue, Dec 23, 2008 at 2:59 PM, Mindaugas Kezys mke...@gmail.com wrote: Looks very interesting. After reading all available info I have two questions before testing: 1. Who/what answers the calls at the other end? I guess real live traffic should be sent through this Asterisk server? 2. How many calls you had made to to diagnose your problems? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions Mindaugas, Thanks. 1) The same Asterisk system places the call and answers the call. You can use one technology to place the call and another to answer (but you don't have to). One connection is the control, the other is the variable. For instance, in my experiments I was testing SIP providers against one another and each other. I placed outbound calls to the SIP provider to a DID I had on a local PRI that came back into the same system running recqual. Recqual then (as designed) played audio in both directions while recording, like this: outbound call (to SIP provider, dialing DID on PRI below): - (audio in from inbound leg) -- (audio out to outbound leg - not recorded, should always be perfect) Inbound call (from PRI): - (audio in from outbound leg) -- (audio out to inbound leg - not recorded, should always be perfect) I know what the audio on both legs should look like so all I need to do is analyze it once it is returned to me. That way I can see how it changed from what Asterisk was supposed to be generating in the first place. This is how recqual is able to detect such a wide range of quality problems. 2) Depends on what you are testing for. For example, if you know you have a bad pair in a bundle of analog lines, you could just call all of the numbers a couple of times. Whichever number/pair has the worst audio (fails the test or looks the worst) is the bad pair. In the case of my SIP provider, they used outbound load balancing to send traffic to multiple media gateways and multiple trunk groups all over the country. Because we had no control over these routing decisions we had to make periodic calls for at least a day to detect the majority of the bad hosts/trunks/etc. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
On Tue, Dec 23, 2008 at 4:02 PM, Atis Lezdins a...@iq-labs.net wrote: Hi, This is good idea, and i will probably try it out someday next year (too busy completing my business requirements :) Luckily next year is just over a week away. We won't have to wait that long ;). I took a look at asterisk patch, and it seems quite simple. I just don't see the point of removing if(debug). You could easily get this additional logging into Asterisk trunk (if preserving RTP info in debug level), and starting asterisk with debug 1. So, then it would be easier to install recqual. Also, being able to run on unmodified version of Asterisk, it would be good to allow keeping current dialplan and just route test calls trough it. So, people would be able to keep track of their billing, etc for those test calls. This is true, however, I wasn't very excited about any other debug messages that might get printed with debug 1. I knew I only needed the endpoint RTP address, so I just removed the if. Of course you could always just run with debug 1 instead of the patch too. Again, this modification isn't strictly required. I just did if for SIP providers that give unpredictable media endpoint IP addresses... :) Also, thanks for showing us magics of ecasound. I have similar project (pbx-test-framework) that allows IVR/Queue/etc testing in automated mode. Recording everything and checking voice quuailty would be great addition :) Ecasound is very, very cool. Recqual is only scratching the surface of what it can do! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: Recqual is collection of scripts using Asterisk and other Linux utilities to measure call quality on an automated basis. -How it Works- Recqual was designed to detect audio quality problems in a call path that may not be visible from the technology being used locally. Whether it's SIP, ZAP, or IAX the fact is there are many potential sources of call quality problems in just about any call being made. Often times a SIP provider may resell services, for example. While the delivery of IP packets to/from this provider may look excellent there may be other problems upstream that an analysis of the IP packets, path, etc may not be able to detect. In scenarios such as this the only way to identify call quality problems is to analyze the audio itself. Regardless of method or transport being used, the goal of any telephony system is to deliver reliable, consistent call quality. Recqual is designed to allow you to place a large number of automated calls (using Asterisk) using different call scenario files. The key here is consistency. When Asterisk places the outbound call (and answers the inbound call) it will generate a set of tones while recording the return audio path. Once the run has finished Ecasound will run with various filters and noise gates to detect certain amounts of distortion, signal loss, etc. Calls either pass or fail based on how much variation there is in the audio once it has been returned. Of course you can pass audio through any combination of networks - including the PSTN. Almost any call quality problem(s) can be detected with this method. Whether it's one way calls, echo, dropped packets, distortion, etc ecasound should be able to isolate the problem calls. If not you can just tweak the script ;). Only calls that fail are saved. These files can be imported into your favorite audio processing utility and/or run through Ecasound again if you'd like to tweak the process script to detect them automatically. Recqual has been designed (and optimized) to work with SIP channels. For example, it has the ability to correlate problem calls with specific RTP endpoint IP addresses. However, due to the protocol independent nature of Asterisk you can use just about any channel type with a few simple changes. --- So there you have it. I've used this with a great deal of success but I think there is still a lot to be done. More on my blog here: http://blog.krisk.org Thoughts? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Application Layer Gateway for SIP protocol
On Fri, Dec 19, 2008 at 2:56 AM, Olfa Echi olfae...@yahoo.fr wrote: Hello everybody, I want to know if Asterisk can provide any solution to perform NAT traversal for SIP protocol which means that it implements the functions of an ALG (Application Layer Gateway). Thanks. Look at the SIP connection tracking module for iptables/netfilter: http://www.calivia.com/iptables-sip-conntrack-nat -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
On Thu, Dec 11, 2008 at 3:19 AM, Shaun Wingrin voi...@gmail.com wrote: Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun nprobe and PF_RING are by far the most comprehensive tools I've seen to do this under Linux: http://www.ntop.org/nProbe.html We've been trying to work something out with Luca (from ntop/PF_RING/nprobe) to further the SIP/RTP abilities of PF_RING/nprobe. We haven't worked anything out yet but I would be interested to hear how the Asterisk community feels about this. The plugin architecture could also allow for an IAX flow analyzer, for instance... I'm also a bit disappointed by the existing flow collectors out there but that's a whole other rant. I can attest the basic claims of performance, speed, and efficiency are all true based on my experiences with nprobe in AstLinux. I don't think I ever fully integrated PF_RING with AstLinux but I understand it increases the performance and capabilities of nprobe dramatically. One of the best features of nprobe is the ability to not only export UDP flows directly to a flow collector but to also write out that data to ASCII and/or binary logs that can later be parsed. If you could combine some timestamps with this flow data you could easily provide for quality monitoring with history for every SIP/RTP (IAX w/ plugin) flow. You could also analyze other flows (HTTP, evil BitTorrent, etc) over the same connection to correlate potential voice quality issues with other types of traffic on the network/circuit. This ability alone is why I think this solution is so powerful. Of course some of these capabilities could be built directly into Asterisk but Asterisk wouldn't give you data on other flows, would it? Also keep in mind a single instance of nprobe/PF_RING running on a Linux router in a large VoIP/Asterisk network could provide flow data and statistics for the entire network (what people do with NetFlow now). Something to think about... Of course another issue is the license and source availability. You have to pay for the source but it's GPL licensed. Let your mind ponder that for a minute... There are some interesting docs, whitepapers, etc on the site (nProbe/PF_RING) if you are interested. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stability unmatched!
On Mon, Dec 8, 2008 at 3:06 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: On Mon, 8 Dec 2008, Danny Nicholas wrote: The 100,000,000 calls without a crash are more impressive to me than the 1000 days of uptime. Mine crashes on crazy things like dynamic conferences, etc. :( To be upfront the system is only running a prepaid AGI app and routing calls for post-paid customers. Can't vouch for more complex setups - all other full fledged PBX systems I have installed go down for other reasons like power. This box has been hosted by The Planet in Dallas, which I cannot recommend enough (no affiliation - just a happy customer). That much uptime at The Planet in Dallas? I guess you're lucky: http://www.thewhir.com/marketwatch/060208_The_Planet_Explosion_Causes_Outages.cfm http://www.datacenterknowledge.com/archives/2008/06/01/explosion-at-the-planet-causes-major-outage/ -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stability unmatched!
On Mon, Dec 8, 2008 at 3:37 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: On Mon, 8 Dec 2008, Kristian Kielhofner wrote: That much uptime at The Planet in Dallas? I guess you're lucky: http://www.thewhir.com/marketwatch/060208_The_Planet_Explosion_Causes_Outages.cfm http://www.datacenterknowledge.com/archives/2008/06/01/explosion-at-the-planet-causes-major-outage/ The explosion was apparently in Houston... Oops! I don't know how I got that confused... Shame on me, and I was just in Texas last week! My apologies. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] directrtpsetup without reinvite
On Mon, Nov 10, 2008 at 8:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I want to be able to bridge two sip channels using direct RTP between my endpoints (Audio IP : not local) but without using reinvites. So I set up my asterisk sip endpoints as follows: [test1] type=friend host=dynamic username=test1 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes [test2] type=friend host=dynamic username=test2 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes ... but it doesn't work. How can I ensure that the RTP is not going through my asterisk box and that the re-invite method is not used? P.S. Both endpoints are using the same codec, so no codec translation takes place. What version of Asterisk is this? Last I heard (from Olle) this option was very experimental and should not be used on production systems. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing the size of voice packets
On Mon, Nov 10, 2008 at 2:40 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: Depends? What is the status of maxptime in Asterisk? ... or the remote end, for that matter... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing the size of voice packets
On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov [EMAIL PROTECTED] wrote: If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? Depends? What is the status of maxptime in Asterisk? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing the size of voice packets
On Mon, Nov 10, 2008 at 4:08 PM, Kevin P. Fleming [EMAIL PROTECTED] wrote: Alex Balashov wrote: If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? Only if the 'ptime' or 'maxptime' values offered are not legal for the codec involved; if they are supported by the codec, then 'ptime' is just a preference, not a demand. The endpoint accepting the offer is free to send packets of any legal size, but not exceeding 'maxptime'. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) Yep, that's how it's supposed to work. Are you confirming our understanding of the spec or Asterisk's implementation of the spec? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing the size of voice packets
On Mon, Nov 10, 2008 at 4:52 PM, Kevin P. Fleming [EMAIL PROTECTED] wrote: Kristian Kielhofner wrote: Are you confirming our understanding of the spec or Asterisk's implementation of the spec? Well the former, and I hope the latter too, since it should match the former :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) Excellent answer! :) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Astlinux-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice Conferencing
On Tue, Nov 4, 2008 at 10:34 AM, Michael Graves [EMAIL PROTECTED] wrote: This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards, device compatibility, etc. The conference call will be held as usual on the Talkshoe service for people calling in from normal (G.711) phones. The Talkshoe bridge can be reached by PSTN or SIP URI. Anyone with G.722 capable phones (some models of Polycom, Snom, Cisco, Avaya, Mitel, Siemens or Grandstream) or a G.722 capable soft phone (Eyebeam, OEM version only) will be able to connect to the ZipDX conference bridge and participate in glorious wideband audio. The two conference bridges will be connected. People connected to ZipDX directly will be able to hear the startling difference that HDVoice makes. This is especially true in conference calls where line quality, accents and background noise all cause intelligibility issues. The downloadable recording of the conference will let everyone hear the difference for themselves. The call will happen Friday Nov 7 at 12 noon EST. To find out more about how to join this call please visit: http://blog.mgraves.org/ or http://voipusersconference.org/ning/ Michael Graves While I appreciate the benefits of wideband audio/G722/etc and your efforts to educate people about it during the VoIP User's Conference, I wince at the thought of hearing a sales pitch. At the very least I find some of this company's existing marketing literature suspect (big surprise there, right): ZipDX is the only audio conferencing service available today that can support the HD Voice capabilities of the Polycom IP 6000 and IP 7000. Our system was designed from the ground up to take full advantage of these devices. You can also experience HDVoice on Polycom's SoundPoint(R) IP-550, IP-560 and IP-650. taken from: http://www.zipdx.com/Info/PolycomSS another gem: Patented No-Codes Conferencing Apparently they've patented dialing out to conference participants and/or reading Caller ID/ANI to bypass the pin. More fantastic work from the US Patent Office! I know there is more to HD Voice than G.722 but I also know that at least (in the FOSS world alone) FreeSWITCH supports conferencing in G.722. Pingtel's sipx uses FreeSWITCH as a conference engine. I would bet with some investigation various other commercial products support it too. This was originally posted to various Asterisk-related groups and I certainly realize FreeSWITCH isn't Asterisk. I just doubt that the CEO from a company that makes questionable claims such as this is going to contribute much to what is (usually) an otherwise interesting discussion about (largely) FOSS technology. More than likely you're going to get a sales pitch for their hosted service, which at $0.10/min per participant is expensive. I understand they offer other conferencing features but it's pretty clear they are leading with their exclusive support for wideband audio. I'm sorry for the sarcastic tone and poor attitude but I'm getting increasingly frustrated with sales and marketing goons infiltrating what are otherwise excellent opportunities for *purely* technical discussions. Asterisk-Biz is overrun, they appear on Asterisk-Users from time to time, and they've all but taken over most conferences. Come on guys, don't let it happen to the VoIP User's Conference!! ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP traffic shaping
On 11/1/08, OCG Technical Support [EMAIL PROTECTED] wrote: This was so interesting I had to move it to its own thread! Is anyone using this script? How does it perform compared to the older WonderShaper script? -M- It was based off Wondershaper originally, enhanced for VoIP traffic and gives the option to use HFSC or HTB. Not only do I use it myself for AstLinux and Star2Star, most of the reports I've (we've) had have been favorable. Try it out! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
On 10/31/08, Jonn R Taylor [EMAIL PROTECTED] wrote: Here is the QOS script that I use on my bridge. http://www.taylortelephone.com/asterisk/astshape You should upgrade to the newer astshape script. It classifies traffic using iptables, which is much more flexible. It also has beta support for the HFSC qdisc: http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/package/iproute2/astshape -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Wed, Oct 29, 2008 at 10:48 AM, Drew Gibson [EMAIL PROTECTED] wrote: Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? We chose to go with a segregated network and certainly don't regret the choice. Voice and data are on separate ports at the desk, avoiding QoS issues completely and reducing confision amongst users who still expect separate Phone and Computer plugs on the wall. The traffic does run through the same switches and inter-switch trunks but always on distinct VLANs. My experience with connecting the desktop computer through the phone has been very poor. Audio breaks up when the computer does large data transfers. Yes, Sir. I'll just look that up in our datab...baba.bas.ss..ss..se In addition our users require gigabit to the desktop. The phones are 100Mb. Worst part is the few Cisco phones we have insist on searching for VLAN (which doesn't exist) for 5 minutes on startup. Hopefully they will be replaced through attrition but despite being over-priced, over-featured and proprietary, Cisco do build robust kit. Sigh. regards, Drew Drew, Disable CDP on the phone and that will go away. I know you said you're not using VLANs but... You can use CDP and set your voice-vlan on Cisco switches. Or... you can install cdp-tools on a Linux box and have it advertise a voice vlan for you! http://gpl.internetconnection.net/ I added the voice vlan support to cdp-tools. ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDP (was Re: network design philosophy and practice)
On Wed, Oct 29, 2008 at 1:28 PM, Drew Gibson [EMAIL PROTECTED] wrote: I tried out the cdp-tools some time ago (it may have been on your recommendation, Kristian) but with no success. Is it possible to disable CDP on the 7940 (image_version : P0S3-08-2-00)? regards, Drew Hmmm... I guess I'd like to know why it didn't work for you but in the meantime it's pretty easy to disable CDP. I've never used 8.x but up through 7.x there was an option to disable CDP in the setup menu on the phone. Because CDP discovery is the first thing these phones do, there isn't a way (at least not one that's practical) to disable it in a config file.* * Classic chicken or the egg... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
On Mon, Oct 27, 2008 at 2:49 PM, Wilton Helm [EMAIL PROTECTED] wrote: Thanks Brendan for the explanation. There is one other idea that struck me, but again, I don't know if it has any merit. My thinking is to keep FAX as FAX and electronic as electronic, rather than introducing a new hybrid approach. Obviously Entering FAX from an electronic source is as old as the FAX modem, and Exiting it electronically is as old as E-FAX, not to mention other alternatives. Is it feasible to simply specify the codec as ulaw or alaw (depending on jurisdiction, I forgot the g numbers) for calls originating from the FXS or whatever the FAX is coming from? Obviously, the bandwidth would be higher in that case, but you can't get around the laws of physics. Yes it is lossy compression, still, but it is the simple, predictable form of lossy compression that the modem in every FAX machine already is programmed to cope with. The only problems I can see would be if the provider who handles the call refuses to accept that codec, or transcodes it to something else. I don't know the likelihood of either of these. Wilton Wilton, Many providers will allow you to do faxing via g711u/g711a (G711u mu-law is used in T countries, G711a a-law is used in E countries). Of course they will allow it - fax modems talk to each other just like we do. They're just doing it with much less tolerance to error and variations in the audio. The provider's gateways, however, should detect the fax tone and disable echo cancellation, etc. What this discussion is forgetting are the issues inherent with packet networks: - latency - jitter - packet loss Standard fax machines communicating via some ATA with a G711u RTP stream cannot correct for these situations. In some severe cases. the modems might not even be able to train. V.x modem standards were not designed for packet networks. For this reason many faxes (especially at higher speeds) will fail (depending on the state of the network) when using a G711*, pass-through, or clear channel codec. You will have a much higher rate of success faxing with G711u over your LAN than a congested cable modem, for instance. That's what T.38 is for. It doesn't even use RTP, it uses UDPTL (UDP Transport Layer) or TCP (rare) to manage the transport of data and correct for transmission errors in various parts of the OSI stack. As we've said before the support for this standard varies and often times just doesn't work. - G711u will fail depending on the condition of the network. - T.38 will fail depending on the type(s) of equipment used. Faxing via VoIP is largely a crap shoot. However, it is important to focus on T.38 because I feel these interop issues can *eventually* be resolved. No one is ever going to fix the issues with packet networks*. That's why they are packet networks. We will have much better luck working towards T.38 interop. * Obviously they are some fixes like MPLS, etc, but that doesn't really help those of us trying to make do with the internet, for example. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?
On 10/27/08, Jeff LaCoursiere [EMAIL PROTECTED] wrote: I'm of the same mind if I cannot find something that works even slightly. I don't suppose there are standard C libraries available :( The world is forcing JAVA on us. j One of the best out there (IMHO): http://sofia-sip.sourceforge.net/ -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sonicwall potentially causing long ping times to SIP phones
On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote: We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 I wouldn't single out SonicWalls when it comes to breaking SIP traffic. Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. I'm looking forward to the day when SIP-TLS is the norm and these devices have no idea what kind of traffic is flowing through them! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How Secure Is Asterisk
On 10/21/08, Eric Chamberlain [EMAIL PROTECTED] wrote: ..snip.. Yes, it's possible to encrypt voice traffic between SIP phones, but there is no standard that works across vendors. ..snip.. Incorrect. SIP TLS/SDES/SRTP works across Cisco/Polycom/Snom/FreeSWITCH/SER family/and more. Asterisk has experimental support for TLS and I know SDES/SRTP is on the roadmap. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How Secure Is Asterisk
On 10/21/08, RE Kushner List Account [EMAIL PROTECTED] wrote: According to the chatter and heated discussions on asterisk-dev, I'd say TCP/TLS is fairly untested and potentially very broken and will remain that way through at least 1.6.1. -Ron I'm on asterisk-dev too. That's why I said experimental. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Problem
On 10/19/08, Ahmed torinto [EMAIL PROTECTED] wrote: After installing a new box and asterisk. i have got these errors [EMAIL PROTECTED] ~]# asterisk Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory mkdir /var/run/asterisk -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-users] asterisk +heartbeat (Wilton Helm)
On Fri, Oct 17, 2008 at 9:29 AM, Steve Totaro [EMAIL PROTECTED] wrote: I have mutihomed boxen on many different networks as well, this has never been an issue. Let's put aside why would you or there is no reason, and then think about it again. Let's just say you wanted two NICs on the same subnet with different IPS, Is it a bug or by design? This whole discussion seems to have forgotten about ARP... The kernel will dynamically learn MAC address to IP address associations as well as which interface the association was learned over using ARP broadcasts. This config is broken. I am fully aware of aggregated (bonding) of links too. I didn't bother to click the link because I assume it is just plain old network bonding (aggregating) like in the Cisco world, you can bond several NICs and get higher bandwidth on a switch, I have three NICS bonded for a three gigabit uplink and that material is too dry for this morning, and if it is what I think it is, I have been doing it for years, let's see I got my CCNA in 97 and renewed sometime or another Most Cisco devices (especially back in the day - 1997?) were using Cisco's EtherChannel: http://en.wikipedia.org/wiki/EtherChannel Which is not quite the same as IEEE 802.3ad (referred to as LACP on some switches). I was working with Cisco devices at the time but I don't remember if I ever had the opportunity to configure bonding on my Cat 2950s... I can tell you that even though 802.3ad is a multi-vendor standard, many Cisco admins still configure EtherChannel between Cisco devices. Whether you are using EtherChannel or 802.3ad the catch is your switch needs to support one or the other and you have to specifically configure switch ports to be a member of that aggregation group. It limits bonded functionality to at least smart switches if not full blown managed switches like those from Cisco, HP, Foundry, etc. With most Linux users being as cheap as they are ;), the Link kernel bonding module provides an ability to bond NICS *without* requiring any special support or configuration on the switch. You are even provided various configuration options at module load time to tweak this. I've never used it (I use 802.3ad) so I can't be exactly sure how it works but I can bet there is some ARP magic in there somewhere... Cisco calls this Multiliink in the router space.. I had three bonded T1s, I could unplug up to two of the T1s and and the internet stayed up up, just at 33% capacity. Depending on how you were doing it (Multilink PPP?) that is VERY different technology. Not to be confused with what we have been calling bonding (sometimes referred to as teaming) which use a variety of Ethernet specific technologies. Although Token Ring, etc might have some equivalent (overlapping?) standards, u - who cares ;)? I am talking about NICs with different IPs on the same subnet. Is Asterisk or Linux deciding to reply to a packet sent to 10.0.0.1 (eht0) by sending that packet through 10.0.0.254? Understood. When you up an interface with an IP address and netmask the kernel automatically inserts a route for that network in the route table (using that interface): ip route show: 10.16.5.0/24 dev eth0 proto kernel scope link src 10.16.5.233 metric 1 default via 10.16.5.1 dev eth0 proto static As you can see I've also added a default route here. Now, if I ping my default route the kernel's ARP cache learns which MAC address that IP has and over which interface: arp -an: ? (10.16.5.1) at 00:13:72:26:36:b7 [ether] on eth0 My guess is that if you had two NICs on the same subnet with different IPs the kernel route table and ARP cache would get pretty confused. This seems so incredibly broken to me I've never tried Something else that seems strange about this arrangement, why would you want to bother to configure other hosts on the LAN differently? You're not really adding bandwidth/reliability (if you could call it that) unless you configure other machines on the LAN to use the different addresses... Weird. In short: If you want to have two NICs on the same network, run them through bonding.ko - PLEASE! ;) If you need other IPs, add an alias to your bonded interface! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER + Asterisk
On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the thing to do. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
On Mon, Oct 13, 2008 at 12:37 PM, Eric Chamberlain [EMAIL PROTECTED] wrote: We're developing the client and don't have control over the server, which may or may not be Asterisk. Adding extra extensions isn't possible. Can OPTION packets be used to verify authentication? -- Eric Chamberlain Most implementations (including Asterisk) don't challenge OPTIONS, at least I don't think they do... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
On 10/10/08, Juan Rodríguez [EMAIL PROTECTED] wrote: After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup media stream for this call. I start getting: ERROR[14844] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error) [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error). I had installed Asterisk-1.4.21, but this version stop from receiving calls after these errors occured. Then I downgrade to version 1.4.19 (because I had have tested that version), but after getting these error it stop from creating the outbound call. The configuration of the * is an incomming call from the my SIP Provider and after internal manage it makes a second call to other destination--DID--. For AGI compatibility issues I could not use Version 1.4.22 (issues whith DeadAGI for billing purpuses). This is my rtp.conf [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=1 rtpend=2 This is my sip.conf for the TRUNK [TRUNK] type=peer nat=never host=destination.public.ip.address fromdomain=my.public.ip.address dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 Please help. -- Juan E. Rodríguez Juan, You might need to increase the number of file descriptors available in Linux. What distro are you on? Are you using the Asterisk startup scripts? In later versions this is done for you automatically if you are running Asterisk as root. Have a look at this: http://www.voip-info.org/wiki/view/file+descriptors -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
On 10/10/08, Juan Rodríguez [EMAIL PROTECTED] wrote: Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Any suggestion?? Here's another (fuller) list, shamelessly lifted from another mailing list: ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 99 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited Make sure these are in your Asterisk startup scripts before Asterisk starts. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
On 10/9/08, Ketema Harris [EMAIL PROTECTED] wrote: Hi I have searched the mailing lists and come across similar threads, but no actual solution. I am trying to use a Cisco AS5300 as a gateway for PSTNr. I have been able to configure it to take outbound calls and send them to the PSTN just fine. Inbound calls however are rejected by asterisk with 488 Not acceptable here code. here are the details: ...lots of details stripped... Thanks for the all of the info. Wading through the various debugs I would guess that Asterisk 1.2's SDP parser does not like the multipart INVITE from the AS5300 with MIME type multipart/mixed, followed by a regular SDP (application/sdp) and GTD (application/gtd) (ew). You have two choices: 1) Disable GTD on the AS5300. I personally don't like GTD and I doubt you need the extra ISUP info anyway. I don't see GTD specifically enabled on the AS5300 but I only have experience with AS5X50s on more recent versions of IOS so there might be a default I don't know about... 2) Update/upgrade to Asterisk 1.4: http://bugs.digium.com/view.php?id=10947 Asterisk 1.4 has much better support for multipart/mixed. While this bug references SIP-T, the underlying problem (multipart messages) is the same. Let us know what happens. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP problems?
Hello everyone, Since I've been working with SIP more and more I've discovered there are still plenty of interop and configuration issues between various pieces of equipment in the real world. I enjoy helping with SIP issues in this forum and others but I thought it would make more sense to aggregate this information in a central location. For instance, earlier today a user had a problem between his Cisco AS5300 and Asterisk 1.2. The solution was fairly technical and not very obvious. I was more than willing to help here but then I thought, wait - what if someone on a Cisco list somewhere has a similar problem? What if I'm not there to read his post and reply? What if he can't find it in the Asterisk archives for some reason? What if he/she never gets the issue worked out? Today I plunked down the $9 for submityoursip.com. My goal is to (eventually) have a site where interop details and implementation quirks between various SIP platforms can be easily searched, discussed, etc. Trying to work with OCS and Asterisk? Need a pointer to a TCP/UDP SIP proxy? Can't figure out how to get your Polycom/Asterisk/Cisco/Snom/Sonus equipment to agree on a codec, method of caller id, or DTMF mode? This wiki should help. I'll be adding some more details, fixing up syntax, etc in the next couple of days but for now I thought I'd get the announcement out to see if anyone would like to help: - Wiki formatting. I don't know anything about MediaWiki. Headings, tables, organization, etc. Help! - SIP devices. Manufacturers, service provider offerings, devices, etc. I've started to make lists to (mostly) empty pages, but this part will never be done! - Debugs/SIP traces. Have a strange interop issue? Post the SIP and we'll (at least I will) take a look at it. Maybe we can figure it out for you and add it to the wiki for everyone. One thing I don't want to do is duplicate effort elsewhere, copy/paste from other sites, etc. If you can link to an external resource, please do! In case you missed it before the address is http://www.submityoursip.com and it's free (of course) and you can sign up for an account if you feel like helping me out... :) Thoughts? Tips? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
On Sat, Sep 27, 2008 at 5:54 PM, Philip Prindeville [EMAIL PROTECTED] wrote: I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of the RTP traffic, and the addresses and port numbers correspond to what's in the SIP INVITE/OK messages (assuming that they don't somehow get munged by NAT after tcpdump looks at them -- there is no NAT device upstream of my Asterisk firewall). I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Thanks, -Philip Philip, We were recently having a few call quality problems with one of our carriers, including very mysterious one way audio on specific pieces of equipment. I created a call generator using Playtones/Monitor to record all four audio paths (only two are important) of a successful call and analyze the resulting recordings with ecasound to detect distortion, one-way audio, audio drops, etc. After several hundred calls we were able to get the carrier to correct the offending pieces of equipment. I'm looking into a way to do this in real time but for now this collection of scripts works pretty well. It's not ready for release but I could get it to you shortly for some testing. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding G.729 files
On Tue, Sep 23, 2008 at 4:44 AM, Alex Balashov [EMAIL PROTECTED] wrote: SOX will do it if you install its G.729 format library. As far as converting a group of files, that's what scripting is for, i.e. for FILE in `find . -type f -name '*.g729'`; do NFILE=$(echo $FILE | sed 's/\.g729/\.wav/g') sox [some args] $FILE ... $NFILE ... done Where can one find this? Is it legal? I don't want to get into all of that... For a known good way to convert (not to mention, %100 legal) you can just use Asterisk. Look at res_convert. Just make sure you have the G729 codec loaded. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
On Thu, Sep 18, 2008 at 12:20 PM, Barton Fisher [EMAIL PROTECTED] wrote: Hi, It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP aware. Apparently, this current firmware/programming is not, one way audio problems. Is there a version that support VoIP directly for this router? Thanks, Bart Bart, IMNSHO, the less SIP aware the better... I have to disable SIP inspection on every IOS/PIX device I come across. Fix the one-way audio problems on your proxy, registrar, etc (in the case, Asterisk). Most SIP ALGs are broken. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
On Thu, Sep 18, 2008 at 4:18 PM, Stefan Gofferje [EMAIL PROTECTED] wrote: Interesting. I have my Asterisk with RFC-1918 IPs behid a NATting PIX and the FIXUP SIP of the PIX makes it very easy for me to use my * as server for external clients as well as as client for SIP providers. The PIX nicely replaces the RFC-1918 IP in the SIP-traffic with the current (dynamic) public IP of itself and keeps track of the RTP traffic. Actually, it also chages the ports in the RTP negotiation and then automatically forward the RTP traffic to the ports, the * was offering. Very very convenient. If the IOS firewall in the newer routers make problems, maybe I should not change to an ISR as I planned :). Terve, Stefan Stefan, Your version of PIX might have finally gotten it right, but even recent 12.4T IOS releases tend to really confuse NAT situations (same seems to go for various PIX releases I've used). Part of the problem might be the use of things like nathelper: http://www.iptel.org/ser/doc/modules/nathelper While not related to Asterisk, inconsistencies across SIP ALGs usually cause various ranges of flags passed to nat_uac_test to fail and/or turn up different results depending on what, specifically, the ALG is doing. NAT handling capabilities at the proxy/registrar, inconsistencies across SIP ALGs, dumb PATs not doing any specific protocol fixups (lowest common denominator), and the increasing use of SIP TLS (no ability to snoop/modify SIP headers or bodies including SDPs) tells me that SIP ALGs are not the best solution in most cases, certainly not long term. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX?
On Thu, Sep 11, 2008 at 8:10 PM, C. Chad Wallace [EMAIL PROTECTED] wrote: At 8:29 AM on 11 Sep 2008, John Millican wrote: Not directly on-topic for this list, but I'd not heard of OpenSIPS before, so I had a look at the website. It looks to be a fork of OpenSER. Does that mean OpenSER development has slowed/ceased, or has the OpenSER project itself morphed into OpenSIPS? Regards, Chris via a quick google:OpenSER is now OpenSIPS www.opensips.org OpenSER continues via OpenSIPS A new name, same project Uhhh, I thought that was Kamailio: www.kamailio.net ...I'm confused. Oh no! While not on-topic for this list the OpenSER thing has been confusing lately. Some company has a trademark on OpenSER. The OpenSER project had to change its name to Kamailio (like the Zaptel-DAHDI issue). Around the same time there were some problems on the Kamailio board. There was plenty of activity on the lists, etc but what I took from it is that Bogdan left Kamailio and forked OpenSIPS. I believe he will continue to commit to both (if they give him commit access to Kamailio back) but OpenSIPS primarily exists for his company (Voice System). Basically if you have a support contract with Voice System you should use OpenSIPS. Otherwise you are free to chose either Kamailio or OpenSIPS for whatever reasons you like. There is no OpenSER anymore. If you are confused you can always just SER (the original from iptel/FOKUS) ;). -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX?
On Tue, Sep 9, 2008 at 3:34 PM, Darren Sessions [EMAIL PROTECTED] wrote: I would suggest using OpenSIPS with Asterisk and bypass IAX all together for this particular application. An OpenSIPS solution will take care of your traveler's NAT issues (and could handle the registrations) while you used Asterisk for voicemail and whatever else. I've personally used this type of general setup in the past with a great deal of success for remote offices and soft-phones on laptops. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ OpenSIPS/Kamailio will only help if the OP doesn't want to wait for Asterisk 1.6 to mature and would like to traverse these firewalls using SIP TLS over a non-standard port (which still may not work) to proxy back to a standard pre-1.6 SIP TLS Asterisk system. Their best bet is to use some type of VPN to traverse these firewalls/NATs. IPSEC, OpenVPN, etc. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SCCP port numbers used for audio stram?
On Fri, Sep 12, 2008 at 11:19 AM, OCG Technical Support [EMAIL PROTECTED] wrote: I have a 7921 wireless phone working with Asterisk, and I want to tighten the wide open port range of my IPTABLES now. I tried allowing only SCCP port (2000) in/out and found that my audio was gone. A quick look at my iptables message shows source port 15886 and dest port 25968 used: FORWARD - Drop: IN=eth1 OUT=eth2 SRC=172.31.253.4 DST=172.31.254.102 LEN=200 TOS=0x18 PREC=0xA0 TTL=63 ID=0 DF PROTO=UDP SPT=15886 DPT=25968 LEN=180 Can anyone tell my 1. which port range I have to open for the audio stream? 2. Is there a way to force SCCP and the phone to use a different port range for audio? Thanks MD SCCP (like SIP, MGCP, etc) uses RTP for audio transport. You will need to modify rtp.conf to change the port range Asterisk uses. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does X-Lite 'remember' Congestion state? (halfway OT)
On Tue, Sep 9, 2008 at 9:42 AM, Paul Schewietzek [EMAIL PROTECTED] wrote: Hi all, I noticed a strange X-Lite behavior, it's connected to an asterisk box. The client registers normally and everything works fine. When I dial out (via E1-PRI) and the called party is unavailable, and asterisk indicates CONGESTION to X-Lite. So far so good. When I try to make another call directly after that (doesn't matter if the same or a different extension is being dialed), X-Lite again tells me about unavailability, but on the asterisk console nothing happens, it seems like X-Lite didn't even try to pass the call to asterisk. The only way to immediately make another call is to restart X-Lite :( After waiting a few minutes, everything works fine again. This behavior is reproducable. I wonder if X-Lite tries to 'remember' about the unavailability, because it thinks 'Hey, we didn't get a connection two minutes ago, chances are we won't get one now', which of course would be stupid when we dial a different extension. Could it be that? Or do you think maybe I'm looking in the wrong direction? Any ideas how to get around that behavior (X-Lite, as far as I can see, has no options available regarding that issue)? Maybe asterisk is able to say 'Don't think you're smart!' to the client phone via SIP? (I don't know much about SIP internals) Kindest regards, Paul We've been experiencing this behavior with CounterPath for a while now. They've acknowledged the bug but haven't provided a fix yet... If you restart the phone or wait about 10 minutes (I think) it should be able to make outbound calls again. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problematic Trunk SIP: Got SIP response 405 Method not allowed
is) on the Samsung to point to your Asterisk system's IP address. -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Semi-OT: ServerBeach for VoIP
Hello, I'm looking at getting a dedicated server from ServerBeach to host some light Asterisk/VoIP/SIP stuff. Has anyone used them for this before? I'm pretty sure I've heard good things (in general) about them but VoIP is a very different animal than web hosting - especially for the network (obviously). ServerBeach uses the Peer1 network which looks pretty good. In fact that's how I found out about them in the first place. Or maybe I can do better than ServerBeach? Does anyone know of a dedicated hosting provider that meets the following specs: - Multiple physical datacenters available by request - Well peered network with multiple Tier 1's (Level3, ATT, Qwest, Verizon Biz, etc) - Dedicated servers running Linux (preferably CentOS) Ideally I'd like to be at $150/mo or less. Bandwidth/peering is important but transfer isn't really an issue - SIP/RTP is just a bunch of small packets! :) Other hardware specs don't matter much either. I'd rather have a Pentium 2 running on an awesome network than have an Athlon 5000 with nothing but dirty bandwidth. Any ideas? -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple asterisk approach
On Mon, Aug 4, 2008 at 2:18 PM, JR Richardson [EMAIL PROTECTED] wrote: Use DUNDi, perfect for this. The protocol is very light, no load on the servers to run it, can handle hundreds of queries a second with no load. You want to use regcontext and a few other things to make it all work together. Here are some papers to guide you: ftp://208.81.55.228/DUNDi_So_Easy.pdf ftp://208.81.55.228/Using_DUNDi_with_a_Cluster_of_Asterisk_Servers.pdf Good Luck. JR JR, Those are great articles. Have you ever considered adding them to the Asterisk Cookbook: http://www.asteriskcookbook.com/wiki/index.php/Main_Page -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experience with Vicidial
On 7/17/08, Matt Florell [EMAIL PROTECTED] wrote: On 7/17/08, Alex Balashov [EMAIL PROTECTED] wrote: Matt Florell wrote: No apologies necessary, I think a lot of what you said is mostly true. Well, thank you. I really appreciate that you're willing to entertain what I am saying without construing it as some sort of attack; it is not in the least bit intended that way. All I can say is, WOW! What could have (possibly) turned into yet another ugly, knock-down, drag-out brawl on Asterisk-Users has turned into a wonderful, positive discussion. Seriously, BRAVO to all of you gentlemen (Alex, Matt, Jay). This thread serves as an example for everyone else: This is how you have a positive, productive discussion. What a breath of fresh air, almost like the Good News portion on the evening news that leaves you feeling all warm and fuzzy... ;) Again, thanks and WELL DONE! -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
On 7/17/08, Steve Underwood [EMAIL PROTECTED] wrote: That is certainly true. Many of the comments people make about codecs owe more to the phone than the codec. However, there are various types of impairment. Even a phone which doesn't sound nearly as good as it could in G.711 mode may still sound much better in wideband mode. A typical $200 business desk phone costs more than a typical mini-Hi Fi set that has a lot more complex bits in it, and sounds fantastic by comparison. Basically business desk phones are a swindle. Steve Do tell! Could someone please explain to me why business desk phones are so expensive? I'm not knocking my friends over at Polycom, Snom, or any other manufacturer but in some cases you can buy a cheap but usable laptop for less than you can buy a phone. What gives? -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to monitor Asterisk logs ?
On 7/15/08, Olivier [EMAIL PROTECTED] wrote: Hi, How can I be notified anytime a given warning message appears in Asterisk logs ? I've got a running system that produces cause 34 warnings (Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)) once or twice a week. I would like to like to be notified (by email, phone, ...) anytime such warning message occurs in log file. I was thinking of using logwatch but wondered if anything better exists. Any advice ? Regards syslog-ng with the proper config file and Asterisk configuration should do the trick: http://www.syslog.org/wiki/Syslog-ng/Syslog-ngWiki -- Kristian Kielhofner http://blog.krisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940 not getting PoE from Linksys SLM224P
On 7/7/08, M B [EMAIL PROTECTED] wrote: Anybody have ideas on how I can troubleshoot? From what I've read cisco VoIP phones should be able to get PoE from these switches. I'm using a straight-through cat5e cable. Plug the phone in and nothing. Is there anyway I can test the PoE switch (it was a refurb unit from CDW) w/out having a PoE device? Thanks- Matt The switch you're using supports IEEE 802.3af standard PoE. The phone (7940) supports pre-standard Cisco in-line power. There are some various hacks for this situation scattered around the internet. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On 7/1/08, randulo [EMAIL PROTECTED] wrote: Hi all, As I mentioned briefly in the SIP takeover thread, I'd like to try to talk about security this coming Friday. I realize it is a holiday in the USA, but do geeks ever take a day off, especially security-conscious geeks? Mark Spencer once said The Bug Tracker is never on vacation!. We will try to start this subject this Friday, but I have no experience at all with this. If you know anyone who is good in this area and would like to share their expertise and talk about security in the asterisk and voip contexts, I'd like to hear from them, especially next Friday July 4th. tia, Randy Randy, I'd love to participate as long as no one minds me calling in from the beach... :) I'm interested in developing my SIP DoS script (and any similar solutions). While I'm reluctant to claim that it or anything like it could protect from a true DoS, it would offer some protection at the application level and that could make all the difference in some instances... As far as wider Asterisk/security issues I think J. Oquendo would be a great guest (hint, hint). -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On 7/1/08, Steve Totaro [EMAIL PROTECTED] wrote: NOT sent from my iPhone or Blackberry very funny, you could add the typed with my thumbs line too. :) I know, although it looks like I'll be waiting in line in a couple of weeks for iPhone 2.0/3G... I just need to remember to update my prefs not to include that in the sig line! As far as your DoS script, do you have a general idea on how the conept would work? Would you just drop the packets from the offending IPs? Sort of... I would make it configurable to allow for dropping, rejecting (using ICMP unreachable for UDP, TCP RST for TCP), and logging. Although I tend to not like anything other than a silent drop for what is attack traffic. I'm not going to be the sucker that's going to saturate my upstream bandwidth by actually responding to DoS traffic. DROP by default, I say. Same thing goes for logging... Using disk I/O and space for attack traffic only makes sense if you've got a properly tuned and dimensioned configuration and/or you're running on a separate box. The hashlimit extension already provides for a /proc interface that (along with standard iptables accounting) could provide for enough information without using something like the LOG target. Ideally you could get really fancy and report the source of these attacks and block them at your upstream (via BGP blackhole or some other means). As long as we remember that any large enough, sophisticated enough DoS/DDoS WILL TAKE YOU OUT unless you have ample resources to deal with it. Even then if one of the larger botnets comes after you, good luck! ;) For security, how about an authentication retry setting in the sip configuration? After X amounts of failed auth or registration attempts, block IP for Y amount of time. It would seem fairly easy to do using realtime with DB entries for IP blocks and expiration. Then a quick query of the same tables would allow an admin to put in permanent rules on a firewall or ACL and also contact that ISP's abuse dept. My main concern with implementing these protections in Asterisk is the expense of starting the thread to deal with the (SIP) traffic in the first place. Although I'm not aware of the specifics, Asterisk reserves a bit of resources for each open SIP channel. Ideally I'd intercept attack traffic in the kernel (or better yet in the kernel on a different machine) before it ever got a chance to use any Asterisk resources in userland. Adding any realtime queries or other DB foo would only serve to amplify the effects of the attack (exceed max number of connections in MySQL, die). Of course the other benefit with a generic Linux solution is the same protections (script) would work for any other SIP application or network device. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts
On 6/30/08, randulo [EMAIL PROTECTED] wrote: Someone should write an asterisk-centric document on this topic, it's likely to become an issue someday. Sounds like a great subject for VoIP USers Conference as well. Any volunteers? iptables string and limit matching could be a start, although I don't really know how well it does with fragments (or if that would even be an issue - especially with UDP). Anyways, it would be cool to develop something with iptables string, limit, and maybe even the Asterisk DB for SIP registries. For instance: - allow unknown addresses to REGISTER/INVITE at a normal rate (10 pkts / minute, or something). Figure that would allow 10 INVITEs (calls) per minute (2 INVITEs per authenticated call). - Allow good addresses (registered from the Asterisk db or previously known good) to pass SIP traffic at a greater rate (maybe even wide open). One could use something unique from the request if they wished - matching on the user agent from the Asterisk SIP DB, for example. This could get tricky... You'd have to be able to look at 407s and INVITEs/REGISTERs with and without nonces to do the job right. It would be neat to do this without having to jump into userland too much in iptables/netfilter. Does anyone want to write a kernel module? ;) -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts
On 6/30/08, David Backeberg [EMAIL PROTECTED] wrote: The thing I was mentioning about hashing addresses is already in the kernel, check out: hashlimit on google, or net/netfilter/xt_hashlimit.c in your favorite 2.6 kernel source The other cases you mention could be done with multiple rules, especially if you know good subnets / netmasks. Exactly. I was thinking something like this (totally untested and raw): # Send the right traffic through our chain $IPTABLES -A INPUT -i $IFACE -m udp -p udp --dport 5060 -j sipdos # INVITE limit $IPTABLES -A sipdos -m string --string INVITE --algo bm \ -m hashlimit --hashlimit $IRATE/minute --hashlimit-mode srcip,dstport --hashlimit-name sip_i_limit -j ACCEPT # REGISTER limit $IPTABLES -A sipdos -m string --string REGISTER --algo bm \ -m hashlimit --hashlimit $RRATE/minute --hashlimit-mode srcip,dstport --hashlimit-name sip_r_limit -j ACCEPT # All other SIP methods... $IPTABLES -A sipdos -m hashlimit --hashlimit $ORATE/minute --hashlimit-mode srcip,dstport --hashlimit-name sip_o_limit -j ACCEPT # DROP everything else $IPTABLES -A sipdos -j DROP It would still be nice to have something a bit smarter (keep track of INVITEs and 407s, for instance) and I don't like using the string match. The all other SIP methods rule is dicey too because of things like OPTIONS, SUBSCRIBE, etc. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2 app_vxml
On 6/27/08, Douglas Garstang [EMAIL PROTECTED] wrote: I just downloaded the app_vxml for Asterisk 1.2 from i6net. Couldn't get it to work. We're using Asterisk 1.2 still, and it looks like the app_vxml binary was linked against libstdc_++-5.x (we have libstdc++-6.x). I grabbed the 1.4 version of the module hoping in vain that would work, but it fails with invalid symbols, which isn't surprising. Any ideas on how I can get this to work? Be nice if i6net provided source! Doug. Doug, Use the 1.2 module and install a libstdc++-compat library. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
On 6/22/08, Michael Graves [EMAIL PROTECTED] wrote: Ok, so now that it's possible to implement SIP over TCP instead of UDP why would I want to do this? Beyond simply integration with M$ OCS. And what are the implications for management of QoS? I would expect that lost packets would be less of a factor. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] Michael, The main advantages for SIP over TCP that I know of (in no particular order): - Better compatibility with NAT devices (it seems some of them don't do UDP well) - Support for TLS - Support for packet fragmentation (to support large/diverse SDPs, headers, etc) I'm sure there are other ones but that's all I can think of this early on a Sunday morning... -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 via MAX TNT ??
On 6/3/08, Joe Carroll [EMAIL PROTECTED] wrote: Quick question for the folks using MAX TNTs for aggregators.. When I send a call out the MAX I get the following…. -- Got SIP response 484 Address Incomplete back from 172.16.10.230 Any ideas on how to make 911 appear as a ten digit number to the device so that it will pass the number out to the PSTN ? I've never used a TNT before but what does your dial pattern matching look like? If you were using Asterisk, your match would probably look like this: _NXXNXX 911 would match NXX but not the remaining digits, hence the 484 Address Incomplete. I bet your TNT is doing something similar However: _NXXNXX _911 Would work just fine (in Asterisk). You need to figure out how to do something similar on your TNT. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
On Mon, Apr 14, 2008 at 3:11 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 02:05:18PM -0500, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? Wow. E911 providers are *municipalities*, aren't they? :-) Could you vague that up a bit, Doug? (Or should I be able to generalize that phrasing into what you actually mean, if I expect to get along here? :-) Cheers, -- jra Wow, that response was completely unnecessary. I think most people (myself included) know what he meant. To actually answer the question - I know many people who have had good experiences with Dash911, now known as Dash Carrier Services (I believe). Good luck Doug! -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
On Mon, Apr 14, 2008 at 3:43 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 03:23:45PM -0400, Kristian Kielhofner wrote: Wow, that response was completely unnecessary. I think most people (myself included) know what he meant. Clearly, no, *I* don't. Or I wouldn't have asked. Everyone else was able to provide a helpful, constructive response. I think that speaks for itself. I think, for my part, that *your* attitude was itself unnecessary. But we can take this off list. I won't bother furthering this, on list or off. I'm fine to leave it as a reminder for all of us: never feed the trolls. I forget about that every once in a while. -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
On Mon, Apr 14, 2008 at 4:45 PM, Adam Moffett [EMAIL PROTECTED] wrote: Ok so did anybody have recommendations? How's 911Enable.com? Anybody have recommendations for a reliable, good valued, E911 provider? I did a while back - Dash 911 / Dash Carrier Services. We looked at 911 Enable. We didn't like their API or they way they handled calls. I don't remember all of the details but you could certainly check out both and see what you think. -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZD Net article
On 4/11/08, Dean Collins [EMAIL PROTECTED] wrote: Lol – though nothing is going to top the Forbes article about Mark Spencer this week http://www.forbes.com/forbes/2006/0410/063.html That article is over two years old... -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISPBX Announces COGOBLUE Interface and PBX Appliances
On 4/3/08, Matt Signorello [EMAIL PROTECTED] wrote: Hi Everyone, My name is Matt Signorello and I'm responsible for wholesale dealers sales here at ISPBX. (www.ispbx.com) Matt, As some others have already pointed out, this list is for non-commercial discussion and you shouldn't have used Tony's existing thread to announce your product. Anyways, I see that ispbx is Asterisk based. What modifications have you made to Asterisk? What other tools/utilities does ispbx use? What license(s) are those tools under? Do you have a download area for the source (Asterisk, etc)? Thanks! -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISPBX Announces COGOBLUE Interface andPBX Appliances
On 4/3/08, John Signorello [EMAIL PROTECTED] wrote: John: CogoBlue is a proprietary software package written by ISPBX. It is not open source. It is currently only available on ISPBX hardware. ISPBX hardware uses Asterisk, probably Linux, and probably dozens (if not more) FOSS applications, libraries, etc. Check out CogoBlue, once you see what a configuration package should be, you may have to reassess what that free software is really costing you. You're new here. This is extremely offensive. Free software gave you and your company a product (ISPBX) and a market (CogoBlue). Where would you be without the free software projects (Asterisk, Linux, etc) ispbx uses? Where would you be without the Asterisk community (hint - you wouldn't have a market for CogoBlue). I'm usually not one to feed the trolls but this comment is over the top. -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft Office Communications Server
On Sun, Mar 9, 2008 at 11:05 PM, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director Matt, I believe OCS only supports SIP over TCP. You'll either need to use Asterisk 1.6/trunk with SIP TCP or install SER/OpenSER as a UDP-TCP proxy. -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft Office Communications Server
On Mon, Mar 10, 2008 at 6:38 PM, [EMAIL PROTECTED] wrote: What is the logic of them using SIP over TCP? Is this a broad industry trend? Or just the latest attempt to get around SIP/NAT issues? Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 I would imagine it's because they plan on doing all kinds of neat stuff with SIP including video, TXT, Windows Updates, who knows... SIP over UDP has some pretty serious packet fragmentation issues. If you end up with a large enough SDP or something that causes a SIP packet to grow larger than the smallest MTU in the path between your two endpoints it doesn't work (no fragmentation support with SIP over UDP). SIP over TCP does not have this problem. Also, you need SIP TCP support for TLS... -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maybe OT: SIP - Missing 407 messages
Hello everyone, I have many Asterisk clients registered to an OpenSER proxy. Sometimes (for reasons unknown) the 407 Proxy Authentication Required sent by OpenSER to Asterisk is not received by Asterisk on the client, causing the call to fail. No other SIP messages or other IP traffic seems to be lost (i.e. no packet loss). We can confirm (via a SPAN session on the switch) that the 407 makes it to the wire. We can confirm (via ngrep/tcpdump) that the 407 is never received by Asterisk. At this point I don't think it's an Asterisk or OpenSER problem but it's so strange I can't be certain. I've already checked everything out on OpenSER with Bogdan and he says it looks good. I believe him :). If anything, I'd like to try the SIP TCP support in Asterisk 1.6 and see if that makes a difference. I hope so... I've done a full writeup on my blog (took advantage of HTML in a couple of places) with a description of the problem. If you think something is relevant, feel free to copy and paste it into your reply for completeness in the archives: ttph://blog.krisk.org/2008/02/missing-sip-traffic.html Thanks! -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verficar VoiceMail
2008/1/14 Gilberto Nunes [EMAIL PROTECTED]: A Monday 14 January 2008 16:01:27, Shane D escreveu: Yes, i can! In fact, I really do! :-) Sorry! thanks Sorry this is an English-only list. Have you tried asteriskbrasil.org? -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Traffic Shaping
On Jan 9, 2008 11:44 PM, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Erik Anderson wrote: On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote: Heh yeah that's what I was thinking of doing. What's the traffic shaping like? Can I specify max bandwidth etc or use hfsc shaping? DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB. Sweet. We had been using HTB but upgraded all our CPE to use HFSC when AstLinux did and found it great. Matt, I've been looking for feedback on HFSC... I don't get to play around as much as I used to and I STILL haven't been able to really compare HTB and HFSC for VoIP traffic. What have you found? How do you like the other changes to AstShape (iptables CLASSIFY, etc)? -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4 with around 230 SIP connections
On Dec 12, 2007 9:41 AM, Russell Bryant [EMAIL PROTECTED] wrote: BJ Weschke wrote: Jerry Geis wrote: Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. I would have some serious reservations throwing this many clients into an app_meetme room which is the foundation layer for the page functionality. Well, it may be ok, especially given that the 230 clients are all marked as listen only. There isn't any mixing going on at all. However, there is almost certainly going to be some lag that you may not be happy with. What happens is that you are spawning 230 threads to make outbound calls and connect them to MeetMe, all at the same time. This process is far from instantaneous. :) I would also be concerned about the effects that this spike in extra processing would have on the quality of any existing calls on the system. But, as with most things, the only way to know for sure is to do some testing. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. Russell, What are your thoughts on SIP/RTP multicast, if any? It's been discussed before. Seems like a great solution for paging (f the phones support it). Anyone interested in a bounty? -- Kristian Kielhofner ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
