Re: [asterisk-users] Just Take dCAP at Astricon?
On Sat, Oct 23, 2010 at 7:36 PM, Paul Belanger wrote: > On Sat, Oct 23, 2010 at 7:07 PM, Steve Totaro > wrote: > > Question: Can I just go to Astricon and take the dCAP exam only? In and > > out? Cost? > > > http://www.astricon.net/dCAP.aspx > Looks like cost is $300. > > > Is the cert on 1.4 or 1.6 now? > > > When I did it in May, I used 1.6.2 for the practical. And I believe > there were 1.6 questions on the written too. > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | > Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger > > Thanks Paul, It seems you must register for something Astricon, the cheapest (same price you quoted) is the Expo only which has the dCAP option for $225 additional. https://www.tmcnet.com/scripts/astricon/astrifall10reg.aspx?theplan=AST-O Thanks, Steve Totaro A guy at TMC has been hounding me since I asked this question through them, he started at $800 and the lowest he would negotiate was $500 and some change. -- _ -- 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] 1.8 Console Welcome Message
On Sat, Oct 23, 2010 at 3:35 PM, Andrew Latham wrote: > Some other people noticed that a few days ago. I think Paul was > looking at it... Here is the thread on asterisk-dev http://lists.digium.com/pipermail/asterisk-dev/2010-October/046697.html -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] Just Take dCAP at Astricon?
On Sat, Oct 23, 2010 at 7:07 PM, Steve Totaro wrote: > Question: Can I just go to Astricon and take the dCAP exam only? In and > out? Cost? > http://www.astricon.net/dCAP.aspx Looks like cost is $300. > Is the cert on 1.4 or 1.6 now? > When I did it in May, I used 1.6.2 for the practical. And I believe there were 1.6 questions on the written too. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] Just Take dCAP at Astricon?
Since it is Saturday evening (7PM EST) I am asking this on the list in case someone who knows sees it and can answer. Astricon is in my back yard for the first time, and I could hit you with a rock. I would always like to attend, and spoke at the 2007 Astricon in Phoenix but don't have the idle cycles. Question: Can I just go to Astricon and take the dCAP exam only? In and out? Cost? I am not a real big cert guy because I have seen too many paper mill cert "boot camps", but due to the proximity, I would definitely take the time to get certified. Is the cert on 1.4 or 1.6 now? Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why such high latency on internal lan?
On Sat, Oct 23, 2010 at 12:31 PM, sean darcy wrote: > My internal lan is small, 100mb, all wired. aastra phones. > > sip show peers > ... > 142/... 10.10.10.42 D A 5060 OK (136 ms) > 144/... 10.10.10.44 D A 5060 OK (138 ms) > 145/... 10.10.10.45 D A 5060 OK (133 ms) > > But pings are < 1ms: > > ping 10.10.10.42 > > rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms > > Why are the sip latencies so high? And is it a problem? And if so, how > do I fix it? > > FWIW, latencies to outside providers over nat are close to ping: > > jnctn/ 5060 OK (7 ms) > teliax/... N 5060 OK (7 ms) > > ping > > rtt min/avg/max/mdev = 3.471/4.120/4.466/0.288 ms > > > sean > > As long as your qualify is set high enough that the phone doesn't become "Unavailable" then your only real worry is if RTP is also experiencing lag, which I highly doubt since your ping times are almost nothing. It is bothersome as an admin, but just realize that it is what it is. Trust the ping times for network latency and show peers for a quick connectivity/registration check. Thanks, Steve T -- _ -- 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] Cepstral voice quality not good
I am using app_swift. As a side note, demo on their website also generates sounds which at places sounds like robotic. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:03 PM, "Darren Sessions" wrote: Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria wrote: > Hello list, > > I hav... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SipSak: Send SIP OPTION with password
It's replying so its up :) On 23 Oct 2010 17:32, "Jonas Kellens" wrote: > Hello, > > I'm trying to use SipSak to check if my Asterisk server is > available/running with the following : > > sipsak -vv -s sip:usern...@sip.domain.tld -c sip:usern...@sip.domain.tld > --password guessthis --hostname XX.XX.XX.63 > > The SIP OPTION is received by Asterisk as follow : > > OPTIONS sip:usern...@sip.domain.tld SIP/2.0 > Via: SIP/2.0/UDP XX.XX.XX.63:36887;branch=z9hG4bK.304f1a46;rport;alias > *From: sip:sip...@xx.xx.xx.63:36887;tag=5e8faf01* > To: sip:usern...@sip.domain.tld > Call-ID: 1586474...@xx.xx.xx.63 > CSeq: 1 OPTIONS > Contact: sip:sip...@xx.xx.xx.63:36887 > Content-Length: 0 > Max-Forwards: 70 > User-Agent: sipsak 0.9.6 > Accept: text/plain > > > and it send back : > > SIP/2.0 404 Not Found > Via: SIP/2.0/UDP > XX.XX.XX.63:36887;branch=z9hG4bK.304f1a46;alias;received=XX.XX.XX.63;rport=36887 > *From: sip:sip...@xx.xx.xx.63:*36887*;tag=5e8faf01* > To: sip:usern...@sip.domain.tld;tag=as29357d12 > Call-ID: 1586474...@xx.xx.xx.63 > CSeq: 1 OPTIONS > Server: Asterisk PBX 1.6.2.10 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Accept: application/sdp > Content-Length: 0 > > > I am not able to change the FROM-header so Asterisk authenticates the > OPTION being sent. > > > Jonas. > -- _ -- 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] Cepstral voice quality not good
Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria wrote: > Hello list, > > I have been using Cepstral's 8KHz voices for my text-to-speech service for > some time now, and have been noticing that the voice quality is really poor, > doesn't matter what phrase I give it to convert. None of the other 8KHz > voices I have ever used were this bad. It doesn't seem good enough system to > be used in a commercial system. Is there any better quality text-to-voice > engine? > Zeeshan A Zakaria > > -- > www.ilovetovoip.com > www.pbxforall.com (beta) > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral voice quality not good
Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice engine? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 IAX Registration
On Sat, Oct 23, 2010 at 3:03 PM, Paul Belanger
wrote:
> Okay, just reproduced your issue and looking at the code now. :)
>
Ok, think I fixed it. You can either apply this patch to 1.8.0, or
svn update the branch I'm working on. Feedback is welcome.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger
Index: channels/chan_iax2.c
===
--- channels/chan_iax2.c (revision 292862)
+++ channels/chan_iax2.c (working copy)
@@ -8700,7 +8700,7 @@
peercnt_modify(0, 0, &p->addr);
/* Stash the IP address from which they registered */
- memcpy(&p->addr, sin, sizeof(p->addr));
+ ast_sockaddr_from_sin(&p->addr, sin);
snprintf(data, sizeof(data), "%s:%d:%d", ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port), p->expiry);
if (!ast_test_flag64(p, IAX_TEMPONLY) && sin->sin_addr.s_addr) {
--
_
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Re: [asterisk-users] 1.8 Console Welcome Message
Some other people noticed that a few days ago. I think Paul was looking at it... ~ Andrew "lathama" Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Sat, Oct 23, 2010 at 4:14 PM, Ryan Wagoner wrote: > With previous Asterisk versions when running asterisk -r a welcome > message is displayed with the version. I just upgraded to 1.8 and > noticed it is not appearing. All I get is Verbosity is at least 3 and > the console prompt. I looked at main/asterisk.c and still see the > welcome message code. Any idea why it is not being shown? > > Ryan > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NAT issues
Hello, this isn't an Asterisk specific problem but I don't know who else to ask for help. This is my setup, it oftens finds double NAT situations: [Asterisk box] <-> [Firewall IPCop] <-INTERNET-> [Random Router] <-> [Softphone] In certain situations, when two or more client softphones use the port 5060 at the same time and try to register, the UDP translation state of the port fails to assure the connection and drops both phones. If I change the client ports to random ones, they register, they can make calls and everything. It just happens if there is port clashing. I am not sure how to tackle this situation as enforcing random ports to the softphones is not viable for the setup. Is this a problem with the IPCop Firewall? I tried flushing the conntrack tables yet this situation kept happening. It gets to the point that nobody can use the 5060 port after a while (when everyone is trying to register). Thank you, Perssy Llamosas. -- _ -- 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] 1.8 Console Welcome Message
With previous Asterisk versions when running asterisk -r a welcome message is displayed with the version. I just upgraded to 1.8 and noticed it is not appearing. All I get is Verbosity is at least 3 and the console prompt. I looked at main/asterisk.c and still see the welcome message code. Any idea why it is not being shown? Ryan -- _ -- 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] a2billing muting "enter the phone number"
If you want to turn off the audio totally you can set audio to NO (it's probably the 4th or 5th in list of Global settings). Otherway is to blank the file responsible to play that file and keeping the settings intact. However, there are numerous options to turn on and off the various announcements which you should look into in the System Settings. -Bruce On Sat, Oct 23, 2010 at 8:31 AM, Baha @ SH wrote: > How can I mute the message "please enter the number you wish to call and > press the # key" in a2billing??? > I tried > use_dnid = YES > but still I keep getting the message prompt... > > thanks > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 IAX Registration
On Sat, Oct 23, 2010 at 1:48 PM, Nic Colledge wrote: > I have just tried the branch you suggested and the problem remains. It's > worse with qualify=yes but still happens (albeit less frequently) with > qualify=no. > Okay, just reproduced your issue and looking at the code now. :) -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why such high latency on internal lan?
>> Why are the sip latencies so high? And is it a problem? And if so, how >> do I fix it? Not a problem at all. Just a goofy Cisco thing. Polycom and Linksys and Grandstream are all a lot lower, but Cisco has always been high. We've seen that for 4-5 years and never had issues. -- _ -- 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] Parinya Sirisang invited you to Dropbox
Parinya Sirisang wants you to use Dropbox to sync and share files online and across computers. Get started here: http://www.dropbox.com/link/20.OHZfc2f_tk/NjQxMzgzNDMwNw - The Dropbox Team To stop receiving invites from Dropbox, please go to http://www.dropbox.com/bl/ee983c10b3d6/asterisk-users%40lists.digium.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 IAX Registration
Paul, Thanks for your reply, I saw that issue on the tracker when I was trying to find a solution earlier but it looked completely different so I didn't give it much thought. This happens on my install with both Realtime phones and those configured in iax.conf I have just tried the branch you suggested and the problem remains. It's worse with qualify=yes but still happens (albeit less frequently) with qualify=no. My iax.conf general section: [general] bindport=4569 bindaddr=0.0.0.0 srvlookup=yes jitterbuffer=no forcejitterbuffer=no context=default rtcachefriends=yes rtautoclear=yes A typicall iax.conf user: [111] type=friend auth=md5 host=dynamic context=internal mailbox=111 accountcode=IAX/111 disallow=all allow=ulaw allow=alaw allow=g729 requirecalltoken=auto qualify=yes secret=secret111 Thanks again, Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 23 October 2010 17:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration On Sat, Oct 23, 2010 at 11:53 AM, Nic Colledge wrote: > Sorry forgot to add this into my initial email. > The same happens with phones configured in iax.conf and the Realtime > database table. > https://issues.asterisk.org/view.php?id=18183 I was able to reproduce a problem with realtime IAX2 yesterday, do you mind trying the branch listed on the issue tracker. Also, please post a copy of your iax2.conf file that will reproduce the issue (mask any passwords). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why such high latency on internal lan?
On 10/23/2010 12:38 PM, Doug Lytle wrote: > sean darcy wrote: >> Why are the sip latencies so high? And is it a problem? And if so, how >> do I fix it? >> > > I've noted that if I run DNS on the Asterisk sever, that my ms times > drop by almost 50% > > Doug > > I hadn't set any dns on the aastra's. I just did pointed to the asterisk box, which is running named. No difference :-( sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 IAX Registration
On Sat, Oct 23, 2010 at 11:53 AM, Nic Colledge wrote: > Sorry forgot to add this into my initial email. > The same happens with phones configured in iax.conf and the Realtime > database table. > https://issues.asterisk.org/view.php?id=18183 I was able to reproduce a problem with realtime IAX2 yesterday, do you mind trying the branch listed on the issue tracker. Also, please post a copy of your iax2.conf file that will reproduce the issue (mask any passwords). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why such high latency on internal lan?
It depends on the type of sip end point, and how long it takes to respond to a SIP event. For example if I connect a Cisco 7960 IP phone to my Asterisk server over the LAN, I always see registration times of over 100ms. But if I connect X-Lite I get registration times of under 10ms. Asterisk connected as a SIP client, under 2ms. The higher latency with the Cisco's don't seem to effect performance at all. -- Brad > To: asterisk-users@lists.digium.com > From: seandar...@gmail.com > Date: Sat, 23 Oct 2010 12:31:58 -0400 > Subject: [asterisk-users] Why such high latency on internal lan? > > My internal lan is small, 100mb, all wired. aastra phones. > > sip show peers > ... > 142/... 10.10.10.42 D A 5060 OK (136 ms) > 144/... 10.10.10.44 D A 5060 OK (138 ms) > 145/... 10.10.10.45 D A 5060 OK (133 ms) > > But pings are < 1ms: > > ping 10.10.10.42 > > rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms > > Why are the sip latencies so high? And is it a problem? And if so, how > do I fix it? > > FWIW, latencies to outside providers over nat are close to ping: > > jnctn/ 5060 OK (7 ms) > teliax/... N 5060 OK (7 ms) > > ping > > rtt min/avg/max/mdev = 3.471/4.120/4.466/0.288 ms > > > sean > > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why such high latency on internal lan?
sean darcy wrote: > Why are the sip latencies so high? And is it a problem? And if so, how > do I fix it? > I've noted that if I run DNS on the Asterisk sever, that my ms times drop by almost 50% Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why such high latency on internal lan?
My internal lan is small, 100mb, all wired. aastra phones. sip show peers ... 142/... 10.10.10.42 D A 5060 OK (136 ms) 144/... 10.10.10.44 D A 5060 OK (138 ms) 145/... 10.10.10.45 D A 5060 OK (133 ms) But pings are < 1ms: ping 10.10.10.42 rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms Why are the sip latencies so high? And is it a problem? And if so, how do I fix it? FWIW, latencies to outside providers over nat are close to ping: jnctn/ 5060 OK (7 ms) teliax/... N 5060 OK (7 ms) ping rtt min/avg/max/mdev = 3.471/4.120/4.466/0.288 ms sean -- _ -- 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] SipSak: Send SIP OPTION with password
Hello, I'm trying to use SipSak to check if my Asterisk server is available/running with the following : sipsak -vv -s sip:usern...@sip.domain.tld -c sip:usern...@sip.domain.tld --password guessthis --hostname XX.XX.XX.63 The SIP OPTION is received by Asterisk as follow : OPTIONS sip:usern...@sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.63:36887;branch=z9hG4bK.304f1a46;rport;alias *From: sip:sip...@xx.xx.xx.63:36887;tag=5e8faf01* To: sip:usern...@sip.domain.tld Call-ID: 1586474...@xx.xx.xx.63 CSeq: 1 OPTIONS Contact: sip:sip...@xx.xx.xx.63:36887 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain and it send back : SIP/2.0 404 Not Found Via: SIP/2.0/UDP XX.XX.XX.63:36887;branch=z9hG4bK.304f1a46;alias;received=XX.XX.XX.63;rport=36887 *From: sip:sip...@xx.xx.xx.63:*36887*;tag=5e8faf01* To: sip:usern...@sip.domain.tld;tag=as29357d12 Call-ID: 1586474...@xx.xx.xx.63 CSeq: 1 OPTIONS Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 I am not able to change the FROM-header so Asterisk authenticates the OPTION being sent. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 IAX Registration
Sorry forgot to add this into my initial email. The same happens with phones configured in iax.conf and the Realtime database table. [Oct 23 16:49:52] ERROR[1220]: chan_iax2.c:8770 update_registry: Bad address cast to IPv4 etc. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 23 October 2010 16:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration What happens without using Realtime ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan help
> On Mon, 18 Oct 2010, Jigar Joshi wrote: > >> @Gilles here are my requirement.can you please help me . On Mon, 18 Oct 2010, Steve Edwards wrote: > Are you putting this "out to bid" or are you just too lazy to read ATFOT > (http://downloads.oreilly.com/books/9780596510480.pdf)? On Sat, 23 Oct 2010, Jigar Joshi wrote: > I am facing issue while generating a dial plan for the following case: > > all caller should be asked a code to enter than All the callers should > be connected one extension. > > also tell me testing scenario : I have pbx setup and currently I have > soft phones to use as extension. > > Currently I have created a dial plan using vdp I tried submitting it > here but I don't know how to extract text version for the same . > > I have deployed that dial plan to my local system and when I dial any > extension call just gets ended. > > At the end I should be able to dial a no from a soft phone and it should > ask me a code then I should be connected to an fixed extension. Have you read the book? Have you tried some of the examples? This "visual dial plan" stuff is leading down a path where you will never understand how a dial plan "really works" and you will not learn how to help yourself -- which you will need to do since nobody on this list seems to know anything about vdp. I'm still trying to understand what you are trying to accomplish. It sounds like you want to allow callers to join a conference after entering a PIN. If so, search the book for examples on using the meetme() application. Google will also prove to be a valuable resource. It also sounds like you haven't mastered even calling from one extension to another. Learn to walk before you try to run. > please also mention how to deploy sound file to system using web > interface. Doesn't your vdp stuff automagically do this for you? Skip looking for some magic visual or web based tools and learn to use the Unix command line -- it's really not all that difficult. If you don't want to invest the time to learn to use the proper tools, please hire someone to do it for you. Do you repair your own [kitchen appliances|plumbing|car|computer]? You can learn any of these skills or you can hire somebody to do it for you. Do you have the basic Unix skills to use cp (from a USB stick), scp, or ftp? Try this approach: 1) Learn enough Unix to log in and edit the Asterisk configuration files using an editor like emacs, vi, or joe. 2) Create a simple dial plan so you can dial a number and play a file like "demo-congrats." 3) Add to your dialplan so you can dial another number and dial another phone. 4) Add to your dialplan so you can dial a number and execute the meetme() application. At each step, observe the console output from Asterisk so you will learn what a normal call looks like and you will see useful messages that will clue you in when something doesn't work as expected. Everybody on this list is interested in helping you succeed with Asterisk, but only if you are willing to invest the effort. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 IAX Registration
What happens without using Realtime ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 IAX Registration
Hi, Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about half an hour ago. IAX Friends (Zoiper Softphones) don't stay registered for more than a few seconds they start out with status unknown and quickly become unreachable, I am using realtime with postgresql, with tables and configuration that have worked fine for IAX in 1.6 and a trunk release from a few months ago. I have not tested any other IAX clients other than Zoiper Softphones. Operating system is Ubuntu 9.10, with the default kernel recompiled for AMD CPUs and a couple of other small changes. I get the following errors on the console after a registration followed by unregistration of the zoiper client: [Oct 23 15:22:50] ERROR[695]: chan_iax2.c:11917 iax2_poke_peer: Bad address cast to IPv4 [Oct 23 15:22:50] ERROR[695]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 [Oct 23 15:22:50] ERROR[695]: chan_iax2.c:8770 update_registry: Bad address cast to IPv4 [Oct 23 15:22:53] ERROR[702]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 [Oct 23 15:22:53] ERROR[703]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 [Oct 23 15:22:53] ERROR[694]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 [Oct 23 15:22:53] NOTICE[702]: chan_iax2.c:8618 reg_source_db: IAX/Registry astdb host:port invalid - '192.168.1.111:4569' [Oct 23 15:22:53] ERROR[693]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 23 15:22:53] NOTICE[703]: chan_iax2.c:8618 reg_source_db: IAX/Registry astdb host:port invalid - '192.168.1.111:4569' [Oct 23 15:22:53] NOTICE[694]: chan_iax2.c:8618 reg_source_db: IAX/Registry astdb host:port invalid - '192.168.1.111:4569' [Oct 23 15:22:53] ERROR[695]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 23 15:22:53] ERROR[695]: chan_iax2.c:2305 peercnt_modify: Bad address cast to IPv4 Is this a configuration issue or something in asterisk? Thanks in advnace. Nic Colledge -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to properly re-configure dahdi
Hi, How to properly re-configure dahdi, when for instance I want to change from TE to NT mode ? I'm planning the following operations : /etc/init.d/asterisk stop /etc/init.d/dahdi stop rmmod dahdi rm /etc/asterisk/dahdi-channels.conf rm /etc/dahdi/system.conf rm /etc/dahdi/modules nano /etc/dahdi/genconf_parameters dahdi_genconf -v modules dahdi_genconf -v system dahdi_genconf -v chandahdi Am I missing something ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem
Hello I am working on TDM2400p. I am having some problems like: when i connect my analog phone with the card there is no dial tone, but i can dial any extension... but after that i can't hear any voice from my receiver i have used different phone sets but still i cant communicate with other extension. Please help me out. Thank you Regards Ali Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan help
Jigar Joshi wrote: > > Currently I have created a dial plan using vdp I tried submitting it > here but I don't know how to extract text version for the same . > After Googling a bit, I found that VDP is Visual Dial Plan for Asterisk. Neat little application, but I doubt you'll find many if any here using it. I also don't agree with their statement: "Why should I use Visual Dialplan? Simply because this is the easiest and fastest way to create Asterisk dialplan. You do not need to have Asterisk dialplan development experience to create large and complex dialplans. Simply drag, drop and connect components to create the dialplan" If you don't have dial plan experience, then when things aren't working, you'll be completely lost. Hence as you are now. I'd suggest you visit http://asteriskdocs.org http://www.voip-info.org And learn how to code a basic dial plan. You'll find many here willing to help you at that point. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- 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] hangup delayed very much on fastagi appliaction of asterisk 1.6
Hello my friends, Previously, we developed fastagi application with Erlang to run on asterisk 1.4, it run very well. But when we try to migrate this application to interface with asterisk 1.6.2.13( the same with 1.6.1 or 1.6.0), We found the hangup issue there, the hangup event will delayed a few minutes when callee hangup , or make an hangup request on cli. The channel be hungup don't release at once on hangup action performed. And on AMI , there is also no hangup event output at once. So my fastagi application hold there too. It'll delay a few minutes until the hangup event output from cli or AMI.we tried dial over dahdi and sip channel, the result is same. This issue don't happen on 1.4, has anyone here experienced the same issue? And share the resolving experiences here ? Is there any special configuration to control this hangup delay behavior? Thank you in advance! Best Regards, Thomas Liu <>-- _ -- 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] B410P - BRI NT 100 Ohm terminator
Hi, My set up is : Asterisk with B410P in NT mode <-cat5 straight cable > Another PBX in TE mode Is the 100 Ohm terminator you can find on B410P boards, necessary when connecting in NT mode to another PBX (set in TE mode) ? Cheers -- _ -- 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] RealTime Voicemail
I am using Asterisk 1.4.36 with Realtime Voicemail from a MySQL database, and whilst I have it all working, I am unable to find a way to customize the content of the email that gets sent to a user when they receive a voicemail. In the past I just edited it in the voicemail.conf file and made the customizations in there, but now that I am using Realtime voicemail from MySQL, my voicemail.conf file has to be an empty file. So does anyone know how it would be possible for me to customize the content of the email, other than hacking the source? Cheers, Brad Hughes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial plan help
Hi, I am facing issue while generating a dial plan for the following case: all caller should be asked a code to enter than All the callers should be connected one extension. also tell me testing scenario : I have pbx setup and currently I have soft phones to use as extension. Currently I have created a dial plan using vdp I tried submitting it here but I don't know how to extract text version for the same . I have deployed that dial plan to my local system and when I dial any extension call just gets ended. At the end I should be able to dial a no from a soft phone and it should ask me a code then I should be connected to an fixed extension. please also mention how to deploy sound file to system using web interface. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
