Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
A similar issue happens to us. Make sure that, for inbound AND outbound calls rtp packets are reaching the other endpoint. If a NAT device(s) is between the endpoints make sure that the device NATs the traffic on BOTH ways (inbound AND outbound). Regards On Saturday 27 September 2008 23:54:37 Philip Prindeville 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 ___ -- 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 ___ -- 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] Vividial issue
On Sun, Sep 28, 2008 at 8:16 AM, Brad [EMAIL PROTECTED] wrote: does anyone have a sample dialplan for vici dial that does not include any pri stuff. I am running exclusively SIP for everything and trying to edit the sample dialplan and removing anything to do with a pri card is becoming a nightmare! Thank you! check in the source there are lot of sample configs shown in the SVN Tree ram ___ -- 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] New User with Calling Card Question
On Sun, Sep 28, 2008 at 3:56 AM, Babcock, Michael Alex [EMAIL PROTECTED]wrote: can a2 billing work on the same system that directadmin is installed? should not be a problem ram ___ -- 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] test call generator
I'm using Sipp to load test, but it lost some SIP message when I increment Call Per Second more than 9. Regards Grey Man escribió: I've used both the Hammer Call Analyzer software and als to the Hammer XMS system which is a server that they install in your rack to do the packet captures and provide you with all sorts of statistics. I suspect the Empirix Hammer products would be able to take care of any load, monitoring or analysis scenarios you have including signalling and media. The price is going to be the issue. When we looked at the solution the Call Analyzer software was 5 figures and the XMS solution was 6. Regards, Greyman. ___ -- 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.7.3/1694 - Release Date: 26/09/2008 06:55 p.m. ___ -- 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] New User with Calling Card Question
thanks; i'm messing with freepbx and what not for the time being. and am going to give that a try. Now, also i'm going to use that asterisk system i have installed on a dedicated box for roleback... smile On Sep 27, 2008, at 10:55 PM, ram wrote: On Sun, Sep 28, 2008 at 3:56 AM, Babcock, Michael Alex [EMAIL PROTECTED] wrote: can a2 billing work on the same system that directadmin is installed? should not be a problem ram ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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] New User with Calling Card Question
i'm using lylix, does anyone know of a good freepbx mailing list? or can i use this mailing list for freepbx questions? mike On Sep 28, 2008, at 12:00 AM, Babcock, Michael Alex wrote: thanks; i'm messing with freepbx and what not for the time being. and am going to give that a try. Now, also i'm going to use that asterisk system i have installed on a dedicated box for roleback... smile On Sep 27, 2008, at 10:55 PM, ram wrote: On Sun, Sep 28, 2008 at 3:56 AM, Babcock, Michael Alex [EMAIL PROTECTED] wrote: can a2 billing work on the same system that directadmin is installed? should not be a problem ram ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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] Conferencing Hardware
We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. Thanks Jim ___ -- 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] G.722 between Eyebeam and a Polycom IP650
Steve Underwood wrote: If I were building a terminal, I'd make mine announce 8000, but accept 8000 or 16000 to try to maximise compatibility. It seems people don't do that. Looking at debug output from 1.6 (using a grandstream), it looks like that is what it does. ___ -- 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] Problem with pickup extension *8 from features.conf using IAX
Hello and thank you for replyes.
Eric, I looked for it on the mailing list and google and did not find something
relevant to be 100% sure that this feature is not supported.
Some information clare I founded in
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups where
it says that for IAX channels I can use the pickup feature from features.conf.
I was looking for an anser to understand if this is supported or not, not to
lose more time trying to make it work.
Shazaum , thank you for your anser, the application Pickup works ok.
My problem is that this application issued from the dial-plan is
directed pickup, thos means that I have to know the exten that is
ringing.
I have difficulties because I an using call queues and the channel is not
anymore only the exten that is ringing, and if I want to pikup a call that is
comming from a queue, I cannot do this with app Pickup(at least I did not find
any way to do this--any help from somebody who did is apreciated.)
Also, since IAX is developed by asterisk, is strange that for SIP there is
support, and for IAX, this kind of application is not supported--this is why I
asked, maybe I am doing something wrong.
In this case(if it is not supportted), shoul we/I open a bug repot to Digium?
Botton line, what i am trying to do is to pickup any call that cames in, direct
call, transfered call, queue call, using IAX, and I am wondering if this is
possible in any way.
Regards,
Cosmin
I believe chan_iax2 does not support call pickup. Search the archives.
Shazaum wrote:
already tested with an exten?
ex:
exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _*8.,n,Hangup()
2008/9/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hello list
I am trying to configure a PBX using Asterisk.
The problem I am havong is the following: I want to use the *8 from
features.conf to pickup any ringing extension from a group, becouse
I want to put the users in call queues and I want anybody from the
company to be able to pick a ringing channel, even if is in a queue.
Whwn using Sip protocol for the users, everithing is going fine, I
can pickup any ringing extension from the group using *8.
But the problem appears when I am using IAX protocol. When issuing
*8 from the IAX phone, asterisk tryes to find the *8 in the dialling
rules returning:
*CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
http://10.0.0.30:4569
[Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process:
Rejected connect attempt from 10.0.0.30 http://10.0.0.30, request
'[EMAIL PROTECTED]' mailto:[EMAIL PROTECTED] does not exist
This I think is wrong, is something like asterisk cannot read from
features.
With the same setting, when using SIP, i get:
*CLI == Using SIP RTP CoS mark 5
[Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092
handle_request_invite: Nothing to pick up for
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
and it works ok.
I am wondering if any had this problem before and if you can help me
figure it out(how to make it work--or if is a bug), or find a
sollution using the app pickup.
I tryed using asterisk 1.4.13, asterisk 1.4.21.2 http://1.4.21.2,
asterisk 1.6-rc6 and always the same problem ocurs.
Regards,
Cosmin
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No virus found in this
Re: [asterisk-users] Dial Plan Issues
no.. it's directly connected to the internet.. it's not an issue of accepting
calls.. see.. the problem is the call gets to the server.. the server tries to
route it..
but as if the dial plan is not there.. it rejects the call because it doesn't
know what to do with it..
for example of my SIP.Conf
[5003]
type=peer
qualify=yes
port=5060
nat=yes
host=HOSTIP
allow=all
dial=SIP/5003
context=from-smarttech
canreinvite=no
call-limit=50
deny=0.0.0.0/0.0.0.0
permit=HOSTIP/255.255.255.255
Extensions.conf
[from-smarttech]
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = s,n,Set(__FROM_DID=${EXTEN})
exten = s,n,Gosub(app-blacklist-check,s,1)
exten = s,n,GotoIf($[ ${CALLERID(name)} != ] ?cidok)
exten = s,n,Set(CALLERID(name)=${CALLERID(num)})
exten = s,n(cidok),Noop(CallerID is ${CALLERID(all)})
exten = s,n,Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}})
exten = s,n,SetCallerPres(allowed_not_screened)
exten = s,n,Goto(ext-queues,8004,1)
let's say smarttech is a voip provider.. which forwards calls to my user on
their system .. now my server is supposed to route those calls according to the
dial plan..
the same exact settings worked like magic on another server.. but on this
server.. it just as if the context and the dial plan does not exist.!!!
any idea?
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Fri, 26 Sep 2008 11:55:45 -0500From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [asterisk-users] Dial Plan Issues
Steve Murphy wrote:
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of
my servers is not allowing the other to send calls while it is possible in the
opposit direction..
i have the same exact settings for the extensions.conf
i tried with another friend of mine.. and connected to his server.. and it
didn't allow him to send me calls..
so my question is..
is it possible that my server is not accepting any context ? it only runs the
ones that come default with Trixbix.. like chanspy, ext-local, from-trunk...
and so on..
what can i do to avoide this problem?? i can't rebuild a new box this one is a
production server and i wasn't making tests.. i was connecting two of my
employer's servers with each other..
regards
Tariq--
You might try a trixbox users mailing list.
There might be a few trixbox users hanging around in
this group who might be able to help, but your
chances are much better in that list.
murf
The server that is not accepting calls is not behind a NAT firewall by any
chance is it?
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Re: [asterisk-users] G.722 between Eyebeam and a Polycom IP650
On Sun, 28 Sep 2008 12:05:06 +0800, Steve Underwood wrote: There is no error in RFC3551. There is a clear statement that an earlier RFC did things wrong, due to a typo, and classifying G.722 as an 8000 sample/second codec is, for better or worse, the standard. Its messy and inconsistent, but its the standard. The only manufacturers I know of who do the wrong thing (i.e. using 16000 in the SDP) are Grandstream and Aastra. Both are aware that their products are incompatible with the rest of the universe, but seem uninterested in fixing them. If I were building a terminal, I'd make mine announce 8000, but accept 8000 or 16000 to try to maximise compatibility. It seems people don't do that. Unless you have some special version of eyebeam, I don't think it supports G.722. It supports G.722.2, but that is completely different. It also supports 16 bit PCM and DVI4 at 16000 samples/second. No, it's the OEM version which I got from ZipDX. The version that Counterpath sells direct to end-users does not have the same selection of codecs as the OEM version. You can see the codec list in the following image from the audio cod dialogue. http://www.mgraves.org/voip/wp-content/uploads/2008/09/eyebeamcodecs.png Are 16 bit PCM and DVI14 supported by any hard phones? Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] Dial Plan Issues
This is a better question asked on a Fonality list. Maybe they have a manual. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. [EMAIL PROTECTED] wrote: Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Want to do more with Windows Live? Learn 10 hidden secrets from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns%21550F681DAD532637%215295.entry?ocid=TXT_TAGLM_WL_domore_092008 ___ -- 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 -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- 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] New User with Calling Card Question
FreePBX has an IRC channel I believe, as well as a forum. I have used ASTCC and ASTPP. Pretty simple, I have never really played with A2billing. Thanks, Steve Totaro On Sun, Sep 28, 2008 at 4:07 AM, Babcock, Michael Alex [EMAIL PROTECTED]wrote: i'm using lylix, does anyone know of a good freepbx mailing list? or can i use this mailing list for freepbx questions?mike On Sep 28, 2008, at 12:00 AM, Babcock, Michael Alex wrote: thanks;i'm messing with freepbx and what not for the time being. and am going to give that a try. Now, also i'm going to use that asterisk system i have installed on a dedicated box for roleback... smile On Sep 27, 2008, at 10:55 PM, ram wrote: On Sun, Sep 28, 2008 at 3:56 AM, Babcock, Michael Alex [EMAIL PROTECTED] wrote: can a2 billing work on the same system that directadmin is installed? should not be a problem ram ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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 -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- 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] Conferencing Hardware
On Sun, 28 Sep 2008, Jim Boykin wrote: We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. I have one server which handles a few simultaneous conferences using just ztdummy - however there are rarely more than 4-5 participants and rarely more than 2 conferences on the go at any one time.. (2.5GHz AMD Semperon FWIW) Ztdummy using: ztdummy: Trying to load High Resolution Timer ztdummy: Initialized High Resolution Timer ztdummy: Starting High Resolution Timer ztdummy: High Resolution Timer started, good to go And zttest gets more 100%'s than not. Gordon ___ -- 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] 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
[asterisk-users] Need help with Cisco 7960
Hi all, This might be a little off topic, but I need some help with this phone and hopefully someone on this list is able to assist me. When establishing a conference call I am not able to hang up the call I connected to my original call. I have tried pressin ghte conference button, but nothing happens. Any help would be apreciated, many thanks! Christian ___ -- 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] Troubleshooting one-way voice... how to peek into SIP RTP?
Well, things just got a lot more interesting... Adding Monitor() to an
extension ends the one-way voice problem on inbound calls!
So an incoming call gets handled as:
[ctc-incoming]
exten = 208345,1,Noop()
exten = 208345,n,Log(NOTICE: RDNIS: ${CALLERID(rdnis)} ANI:
${CALLERID(ani)})
exten = 208345,n,Goto(redfish-pstn,s,1)
...
[redfish-pstn]
exten = s,1(incoming),Noop()
exten = s,n,Answer()
exten = s,n,Wait(0.5)
...
some filters for bogus ANI's like 8 goes to badani below
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,nWaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
exten = s,n(badani),Log(DEBUG,ANI: ${CALLERID(ani)} clearing)
exten = s,n,Playback(privacy-unident)
exten = s,n,Wait(0.5)
exten = s,n,Congestion()
exten = s,n,Hangup()
include = redfish-extens
exten = i,1,NoOp(Invalid: ${EXTEN})
exten = i,n,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
[redfish-extens]
...
exten = 113,1,Monitor(wav,,w); for debugging
exten = 113,n,Macro(stdexten,113,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
...
exten = 113,1,Macro(stdexten,119,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
So I don't get this at all. If I dial 208345, then enter '119' as
the extension, it rings on a few phones (including a Xlite softphone)
and if I pick up on any of those, I get one-way voice (I can hear the
caller but they can't hear me).
If I enter '113' as the extension, it rings on two SPA-942's (one of
which is the same as above, just a different line presentation)... and
if I answer, then I get two-way voice! Only difference is the Monitor()
statement.
I'm starting to suspect it's a CODEC issue in Asterisk, though (a) why
Asterisk would need to transcode a call between two uLaw endpoints, I
don't know... and (b) why is it staying in the Media path at all?
I have the SIP peer that the calls come in on as:
[sip-proxy]
...
type=peer
nat=no
canreinvite=no
reinvite=no
Anyone know why the Monitor() would change the duplex(ity) of the audio
stream? I'm baffled (no pun intended). And is there any debugging I
can turn on to reveal CODEC behavior that might differ from 113 and 119?
Thanks,
-Philip
Philip Prindeville 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
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Re: [asterisk-users] Dial Plan Issues
this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sun, 28 Sep 2008 10:00:56 -0400From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [asterisk-users] Dial Plan Issues This is a better question asked on a Fonality list. Maybe they have a manual.Thanks,Steve Totaro On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. [EMAIL PROTECTED] wrote: Greetings,i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction..i have the same exact settings for the extensions.confi tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls..so my question is..is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on..what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other..regardsAHD Tarek SawahIntegrated Digital SystemsCCNA, MCSE, RHCE, VoIPSyria: +963 944 618286USA: +1 347 562 2308_Want to do more with Windows Live? Learn 10 hidden secrets from Jamie.http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008___-- Bandwidth and Colocation Provided by http://www.api-digital.com --AstriCon 2008 - September 22 - 25 Phoenix, ArizonaRegister Now: http://www.astricon.netasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Thanks,Steve Totaro1.888.777.18881.240.938.1212 (cell) _ See how Windows Mobile brings your life together—at home, work, or on the go. http://clk.atdmt.com/MRT/go/msnnkwxp1020093182mrt/direct/01/___ -- 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] Dial Plan Issues
On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! What is the dialplan? ls -ld /etc/asterisk /etc/asterisk/extensions.conf And what is the contents of extensions.conf ? What is the output of 'dialplan show' from the CLI? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Troubleshooting one-way voice... how to peek into SIP RTP?
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? Use tcpdump to capture to a file both call scenarios. Then use Wireshark to open the file. You can then do an 'RTP- Show All Streams' Analysis of the calls. That alone would reveal whether the Audio is really there or not. You can export that G711 Payload and listen to it with the Windows Media Player. If you don't see the RTP in one direction then you might have a signalling problem. Andres http://www.neuroredes.com Thanks, -Philip ___ -- 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 ___ -- 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] Problem with my softphone
Hello, when with my client X-lite try to register in the server that say me, Registration error:501 Not implemented. What isn't implemented? the registration in the sip.conf or extensions.conf? how can i implemented that? thanks. Abel ___ -- 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] Conferencing Hardware
Go for it. ztdummy is not an issue. I have used ztdummy with 220 simultaneous participants in 18 different conference groups. At one time, I had 60 machines running simultaneously in a FARM all of which were carrying the same 18 conference groups with over 200 participants active on each machine. ..mike.. At 11:23 AM 9/28/2008, Gordon Henderson wrote: On Sun, 28 Sep 2008, Jim Boykin wrote: We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. I have one server which handles a few simultaneous conferences using just ztdummy - however there are rarely more than 4-5 participants and rarely more than 2 conferences on the go at any one time.. (2.5GHz AMD Semperon FWIW) Ztdummy using: ztdummy: Trying to load High Resolution Timer ztdummy: Initialized High Resolution Timer ztdummy: Starting High Resolution Timer ztdummy: High Resolution Timer started, good to go And zttest gets more 100%'s than not. Gordon ___ -- 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 ___ -- 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] Conferencing Hardware
Thanks Gordon Mike for the response. What accuracy are you getting from zaptest/dahdi_test (and system info). Two more questions: 1) Does ztdummy requires change into kernel? I am running 2.6.9 kernel. 2) What about CPU load? Thanks Jim On Mon, Sep 29, 2008 at 5:02 AM, Mike Trest [EMAIL PROTECTED] wrote: Go for it. ztdummy is not an issue. I have used ztdummy with 220 simultaneous participants in 18 different conference groups. At one time, I had 60 machines running simultaneously in a FARM all of which were carrying the same 18 conference groups with over 200 participants active on each machine. ..mike.. At 11:23 AM 9/28/2008, Gordon Henderson wrote: On Sun, 28 Sep 2008, Jim Boykin wrote: We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. I have one server which handles a few simultaneous conferences using just ztdummy - however there are rarely more than 4-5 participants and rarely more than 2 conferences on the go at any one time.. (2.5GHz AMD Semperon FWIW) Ztdummy using: ztdummy: Trying to load High Resolution Timer ztdummy: Initialized High Resolution Timer ztdummy: Starting High Resolution Timer ztdummy: High Resolution Timer started, good to go And zttest gets more 100%'s than not. Gordon ___ -- 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 ___ -- 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 ___ -- 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
