Re: [asterisk-users] asterisk 1.6.1/1.6.2 binary packages
Wonderfull ;) On Mon, Apr 5, 2010 at 7:58 PM, Jason Parker jpar...@digium.com wrote: bruce bruce wrote: Thanks for the update Jason, How do the upgrades work if v1.6.0 is already install and one wants to upgrade to 1.6.2 (once it's available)? yum upgrade asterisk* ??? Thanks It should be as easy as a `yum update`. That's the goal, anyways. -- _ -- 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] Which rule for Asterisk to Asterisk-addons compatibility ?
Hello, In asterisk-addons-1.6.1.2's changelog, you can see that 6 changes were committed between versions 1.6.1.1 and 1.6.1.2. But if I'm not mistaken, you cannot read anything there about Asterisk to Asterisk-addons compatibility. What is the rule for Asterisk to Asterisk-addons compatibility ? Is this rule implicit (any Asterisk-addons 1.6.1.X is compatible with any Asterisk 1.6.1.Y) or did I miss 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
Re: [asterisk-users] OT: Wireless headset / phone combination
On Mon, 5 Apr 2010, Warren Selby wrote: On Mon, Apr 5, 2010 at 9:37 PM, Alec Davis siva...@paradise.net.nz wrote: I've been asked for recommendations for a small call centre, an ethernet SIP deskphone with a wireless headset. Similar approach would be a mobile phone with bluetooth head set. Either I've not looked hard enough, or there isn't much on offer. Alec Davis I guess the main question would be how much are you willing to spend? You can get some good wireless headsets from Plantronics for around $200 (includes the headset, base, and lifter for the phone). I have a client that uses several CS55 headsets with the HL10 lifter and they're very happy with them. Seconded - I've clients with the Plantronics CS60 and CS70's with the HL10's on a combination of Snom and Grandstream GXP2000 desk phones. A bit extra desk spaghetti, but they think it's worth it... Plantronics also have a USB cordless headset too - for use with a soft-phone (although I'm not sure how it sends the 'answer' signal back to the PC/softphone) Another client of mine uses Siemens DECT phones with a wired headset to the phone (standard 2.5mm jack - so most mobile headsets work) clip the phone to their belt (or put it in their pocket) and off they go.. It's a cheaper solution to the Plantronics at the expense of a bit of fiddle factor... Gordon -- _ -- 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] Anyone coming to Paris next week for AstriEurope?
Several regulars from the VUC will be there, some of us are arriving Tuesday night. Anyone else considering the trip? Post here or contact me off list so we can meet. /r -- _ -- 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] OT: Wireless headset / phone combination
snip Seconded - I've clients with the Plantronics CS60 and CS70's with the HL10's on a combination of Snom and Grandstream GXP2000 desk phones. A bit extra desk spaghetti, but they think it's worth it... /snip Seems like it's either 2 or 3 devices to make this work. The lifter is not required, as mostly operators are sitting at stations, wired headsets ensures this :) I had in mind something like the Zultys ZIP 4x5 IP Phone and it's bluetooth headset. But they're old now. And I do wonder, how long do the batteries last on a charge, a call centre operator could be talking for a good period of the day. Alec Davis -- _ -- 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] Anyone coming to Paris next week for AstriEurope?
Randy R a écrit : Several regulars from the VUC will be there, some of us are arriving Tuesday night. Anyone else considering the trip? Post here or contact me off list so we can meet. /r I'll be there but I don't know exactely when 'cause I'll at Paris this week for my Microsoft course -- Cordialement, / Greetings, Georghy FUSCO -- _ -- 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] Anyone coming to Paris next week for AstriEurope?
I'll be there but I don't know exactely when 'cause I'll at Paris this week for my Microsoft course If you're on Twitter, follow @voipusers if you want to keep in touch or email me if you prefer. /r -- _ -- 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] polarity reverse
Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no answeronpolarityswitch=yes I use asterisk 1.6.2 and sangoma a400 fxo ports. Then i try call i get chan_dahdi.c: Ignore possible polarity reversal on line seizure -- _ -- 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] Exceptionally long voice queue length errors...
James Lamanna wrote: I'm seeing a lot of Exceptionally long voice queue length errors in my logs, and then I seem to have a problem where Asterisk will drop the registration for a significant number of phones (they go UNREACHABLE), but then they come back approximately a minute later. Is this some sort of load problem? Or something else? Are you using SIP or IAX? This sounds like issue 15609 which has been resolved in newer versions of Asterisk: https://issues.asterisk.org/view.php?id=15609 I'd try upgrading Asterisk to see if that resolves the problem (assuming you're using chan_sip). There is also an open issue with a similar problem for what looks to be related to IAX2 here: https://issues.asterisk.org/view.php?id=16507 Leif. -- _ -- 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] SIP Dialplan Failover Solution
Hello list,
I need a hand to find the best dialplan failover solution when
using two SIP Trunks.
My reasons to do failover are:
a) one of the two providers could be unreachable
b) both providers may be UP but one of them could return a SIP error
message (maybe caused by DOWN E1s)
Googling I found a few possible solutions:
1. Using DIALSTATUS variable.
2. Dialing in sequence:
exten = _X.,1,Dial(SIP/${TRUNK1}/${EXTEN})
exten = _X.,2,Dial(SIP/${TRUNK2}/${EXTEN})
3. ChanIsAvail
Using the first method it's possible to get the CONGESTION and
CHANUNAVAIL status which pretty much solves my problem but it takes more than 2
lines of dialplan(I like one liners).
The second solution requires less space in the dialplan but it should work only
when the called party is busy (or maybe even when the first trunk is down).
Is there a clean way to send the call to the second SIP provider if the first
one is unreachable or spits out sip error messages?
Thanks in advance,
Alex
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[asterisk-users] Asterisk and MWI with Exchange 2010
I have been working on getting Asterisk and Exchange 2010 UM working together, and so far I am pretty happy. The one thing not working right now is MWI. I searched a bit and found this: https://issues.asterisk.org/view.php?id=13028 Now, please pardon me for being ignorant of all of this, but I am trying to figure out if this has been actually implemented yet. It looks like it has, but only in the 1.6.2 branch, is this correct? I am running 1.6.0.26, but would be willing to move to the 1.6.2 branch if this feature will never be released in 1.6.0. Thanks. -Jay -- _ -- 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] Limit Number Of Simultaneous Outbound SIP Calls
Is there a way to limit the number of simultaneous outbound SIP calls made by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but that doesn't seem to be working and one of our engineers says that parameter has been depreciated. Thanks, deric.p...@nisc.coop -- _ -- 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] Limit Number Of Simultaneous Outbound SIP Calls
Here's one way - put your dial command into a macro that polls via a core
show channels and only dials when the count is below X. Even using a
slow language like PHP or PERL, an AGI call/return would not add as much
time to the dial process as PSTN delay does.
Example:
- exten = 100,1,noop(check before dialing)
- exten = 100,n,AGI(howmanycalls.agi)
- exten = 100,n,Gotoif(${ACTIVECALLS} 10?dial:congest)
- exten = 100,n(congest),play(congest)
- exten = 100,n,hangup
- exten = 100,n(dial),Dial(SIP/${EXTEN}
- exten = 100,n,hangup
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deric Page
Sent: Tuesday, April 06, 2010 8:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Limit Number Of Simultaneous Outbound SIP Calls
Is there a way to limit the number of simultaneous outbound SIP calls made
by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but
that doesn't seem to be working and one of our engineers says that parameter
has been depreciated.
Thanks,
deric.p...@nisc.coop
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Re: [asterisk-users] SIP Dialplan Failover Solution
Hi,
I do use the first solution based on DIALSTATUS variable. (
http://www.voip-info.org/wiki/view/Superdial+macro)
since it's included to a separated context named [superdial-macro], I don't
have to repeat it over and over, so the fact that it's not a oneliner
doesn't bother me at all :)
On Tue, Apr 6, 2010 at 3:37 PM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
Hello list,
I need a hand to find the best dialplan failover solution
when using two SIP Trunks.
My reasons to do failover are:
a) one of the two providers could be unreachable
b) both providers may be UP but one of them could return a SIP error
message (maybe caused by DOWN E1s)
Googling I found a few possible solutions:
1. Using DIALSTATUS variable.
2. Dialing in sequence:
exten = _X.,1,Dial(SIP/${TRUNK1}/${EXTEN})
exten = _X.,2,Dial(SIP/${TRUNK2}/${EXTEN})
3. ChanIsAvail
Using the first method it’s possible to get the CONGESTION
and CHANUNAVAIL status which pretty much solves my problem but it takes more
than 2 lines of dialplan(I like one liners).
The second solution requires less space in the dialplan but it should work
only when the called party is busy (or maybe even when the first trunk is
down).
Is there a clean way to send the call to the second SIP provider if the
first one is unreachable or spits out sip error messages?
Thanks in advance,
Alex
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Mvh,
Aurimas Skirgaila
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[asterisk-users] IAX Problem
I have2 Trixbox Servers. Each has an IAX trunks to the other. One works the other fails: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/526-09eec7c8, IAX2/InterOffice/210,300,tr) in new stack -- Called InterOffice/210 -- Hungup 'IAX2/InterOffice-7578' == Everyone is busy/congested at this time (1:0/0/1) The only difference I am aware of is that one server has a public IP address, the other is behind a NAT. The trunk from the server with the public address works fine. I added nat=yes to the other's peer details - did not help. What should I do? -- Bob Gailer 919-636-4239 Chapel Hill NC -- _ -- 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] Cache sound files for faster processing
Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk On Mon, 5 Apr 2010, Luki wrote: Not directly, but it's not really needed. A long as the machine has enough RAM, the files will be served from RAM by the operating system. Sure there is the overhead of opening/closing files and reading them, but on modern OS this overhead is negligible if the files are cached (asterisk may even use mmap, but I'm not sure). You can also make a ram disk (say via tmpfs), copy the sounds there and symlink the sound directory to that location. However, I don't think you will gain much. A bit off topic, but recently I was trying to improve the performance of a MythTV frontend (a Linux home theater application). I tried tmpfs and /dev/ramx and neither yielded noticeable improvement. My informal conclusion is that Linux does a good enough job at managing memory that tweaking is probably not worth it. -- 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] Limit Number Of Simultaneous Outbound SIP Calls
On Tue, 6 Apr 2010, Deric Page wrote: Is there a way to limit the number of simultaneous outbound SIP calls made by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but that doesn't seem to be working and one of our engineers says that parameter has been depreciated. How about using GROUP() and GROUP_COUNT()? -- 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] IAX Problem
On Tue, 6 Apr 2010, bob gailer wrote: I have2 Trixbox Servers. Each has an IAX trunks to the other. One works the other fails: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/526-09eec7c8, IAX2/InterOffice/210,300,tr) in new stack -- Called InterOffice/210 -- Hungup 'IAX2/InterOffice-7578' == Everyone is busy/congested at this time (1:0/0/1) The only difference I am aware of is that one server has a public IP address, the other is behind a NAT. The trunk from the server with the public address works fine. I added nat=yes to the other's peer details - did not help. What should I do? 0) Use a more descriptive subject. 1) Set auth=plaintext (only for the duration of the debug session) and enable iax2 debugging on the CLI. You should see the NEW request and the passwords in plaintext. Verify the username, password, context, and extension all exist. 2) Reply with the sanitized console output. -- 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] OT: Wireless headset / phone combination
On Tue, 06 Apr 2010 20:49:50 +1200, Alec Davis wrote: snip Seconded - I've clients with the Plantronics CS60 and CS70's with the HL10's on a combination of Snom and Grandstream GXP2000 desk phones. A bit extra desk spaghetti, but they think it's worth it... /snip Seems like it's either 2 or 3 devices to make this work. The lifter is not required, as mostly operators are sitting at stations, wired headsets ensures this :) I had in mind something like the Zultys ZIP 4x5 IP Phone and it's bluetooth headset. But they're old now. Yes, old and unsupported. These have not been manufactured for quite some time. I had one and thought it very promising but the hardware quality was not what I had hoped. And I do wonder, how long do the batteries last on a charge, a call centre operator could be talking for a good period of the day. I might have hoped that snom would use one of the USB ports on the 820 or 870 models to provide support for a Bluetooth headset. The Cisco SPA-525G supports Bluetooth, as does the Aastra 6739i. Both are higher end models. Myself I rely upon a Counterpath soft phone and use the Plantronics Savi Go, which comes with a Class 1 Bluetooth USB dongle. It works great with my desktop and my cell phone. -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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 send calls to SIP Trunks with Round Robin Call Distribution
I have a special requirement that insist an Asterisk server, 1.6.1.x,
is used.? I will have 2 SIP trunks coming into the server and I will
have to send calls to these SIP trunks with a round robin distribution
pattern.? I was thinking of using a group count function, if call
count is even send call to SIP Trunk 1, if call count is odd, send
call to SIP Trunk 2.
The decimal portion of ${UNIQUEID} is incremented every time Asterisk
creates a channel. Applying your even/odd logic to this should work
fine.
Thanks Steve, works great:
exten = _X.,1,Set(uniqueidcut=${CUT(CDR(uniqueid),.,2)})
exten = _X.,n,Set(result=${MATH(${uniqueidcut}%2)})
exten = _X.,n,GotoIf($[${result} 0 ]?siptrunk1,1:siptrunk2,1)
I don't have any empirical evidence, but I would suspect a variable
reference (${UNIQUEID}) would be insignificantly faster than invoking a
function that references a variable (CDR(uniqueid)).
Also, for the forseeable future, the Unix epoch will be 10 digits, so I
suspect specifying the character offset in the variable reference (:11*)
will be insignificantly faster than invoking a function (${CUT()}).
And, unless you have a specific need for the decimal portion of the
UNIQUEID, you could roll it all into a single conditional like:
gotoif($[${MATH(${UNIQUEID:11} % 2,int)} 0]?siptrunk1,1:siptrunk2,1)
*) Assuming you're not using the systemname prefix.
I believe you are correct, that would be more efficient.
After mocking this up, the results were not as expected. The even/odd
modulus worked fine using the ${UNIQUEID}, it actually worked too
well. The issue I ran into was each inbound call was consistently
even or odd so all calls went to the same outbound trunk. Each call
would initiate another SIP call out, so the counter would do exactly
what it is supposed to do and increment on each SIP channel. It seems
pretty obvious now that I think about it. So the call distribution to
the outbound trunks will not work based on the incrementing counter of
the ${UNIQUEID}.
After some thought, I decided to send all outbound calls through a
GROUP_COUNT function and distribute calls to the trunks based on
grater-than GottoIf statement like this:
[inbound]
exten = _X.,1,GotoIf($[${GROUP_COUNT(siptrunk1calls)}
${GROUP_COUNT(siptrunk2calls)}
]?siptrunk2,${EXTEN},1:siptrunk1,${EXTEN},1)
[siptrunk1]
exten = _X,1,Set(GROUP()=siptrunk1calls)
exten = _X,n,Dial(SIP/${ext...@siptrunk1,60,)
[siptrunk2]
exten = _X,1,Set(GROUP()=siptrunk2calls)
exten = _X,n,Dial(SIP/${ext...@siptrunk2,60,)
This worked as expected and is evenly distributing inbound calls to
both SIP trunks based on channel usage, with is ultimately desired.
Of course this is not exactly a round robin distribution but works for
what I need.
Thanks.
JR
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Engineering for the Masses
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Re: [asterisk-users] IAX Problem
Thank you for your interest in my question and quick response. I am relatively new to Asterisk, so I have a few specific questions regarding your suggestions. Then I will post to the list with a more meaningful subject and results. On 4/6/2010 10:31 AM, Steve Edwards wrote: On Tue, 6 Apr 2010, bob gailer wrote: I have2 Trixbox Servers. Each has an IAX trunks to the other. One works the other fails: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/526-09eec7c8, IAX2/InterOffice/210,300,tr) in new stack -- Called InterOffice/210 -- Hungup 'IAX2/InterOffice-7578' == Everyone is busy/congested at this time (1:0/0/1) The only difference I am aware of is that one server has a public IP address, the other is behind a NAT. The trunk from the server with the public address works fine. I added nat=yes to the other's peer details - did not help. What should I do? 1) Set auth=plaintext (only for the duration of the debug session) Where / how do I do that. Is that in the trunk peer settings? enable iax2 debugging on the CLI. I did that; I now get without any action on my part: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 00714 DCall: 0 [67.228.218.114:4569] Whence cometh those lines? When I call I get: VERSION : 2 CALLED NUMBER : 210 CODEC_PREFS : (ulaw|alaw|gsm) CALLING NUMBER : 526 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Bob's Office LANGUAGE: en FORMAT : 4 CAPABILITY : 14 ADSICPE : 2 DATE TIME : 2010-04-06 10:54:08 You should see the NEW request and the passwords in plaintext. What passwords? Where / how does one specify passwords? Verify the username, password, context, and extension all exist. 2) Reply with the sanitized console output. What do you mean by sanitized. I assume you want just the relevant output. True? Thanks in advance. -- Bob Gailer 919-636-4239 Chapel Hill NC -- _ -- 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 and MWI with Exchange 2010
Jay Vocaire wrote: I have been working on getting Asterisk and Exchange 2010 UM working together, and so far I am pretty happy. The one thing not working right now is MWI. I searched a bit and found this: https://issues.asterisk.org/view.php?id=13028 Now, please pardon me for being ignorant of all of this, but I am trying to figure out if this has been actually implemented yet. It looks like it has, but only in the 1.6.2 branch, is this correct? I am running 1.6.0.26, but would be willing to move to the 1.6.2 branch if this feature will never be released in 1.6.0. Features are not backported to previous branches. If the feature does not exist in the 1.6.0 branch but does in the 1.6.2 branch, then you will need to upgrade. Information about 1.6.x versioning can be found here: http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ As a follow up to that, we're moving away from the version number styles we implemented for 1.6.x branches. Information about why that is can be found here: http://blogs.asterisk.org/2010/01/29/about-asterisk-1-6-2-release/ Leif. -- _ -- 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 and MWI with Exchange 2010
On Tuesday 06 April 2010 07:46:05 Jay Vocaire wrote: I have been working on getting Asterisk and Exchange 2010 UM working together, and so far I am pretty happy. The one thing not working right now is MWI. I searched a bit and found this: https://issues.asterisk.org/view.php?id=13028 Now, please pardon me for being ignorant of all of this, but I am trying to figure out if this has been actually implemented yet. It looks like it has, but only in the 1.6.2 branch, is this correct? I am running 1.6.0.26, but would be willing to move to the 1.6.2 branch if this feature will never be released in 1.6.0. Features not in a particular release branch will never be added to that release branch, as a matter of policy, unless the feature is required to fix a security issue, and the release branch is still supported for security issues. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which rule for Asterisk to Asterisk-addons compatibility ?
On Tuesday 06 April 2010 03:16:45 Olivier wrote: In asterisk-addons-1.6.1.2's changelog, you can see that 6 changes were committed between versions 1.6.1.1 and 1.6.1.2. But if I'm not mistaken, you cannot read anything there about Asterisk to Asterisk-addons compatibility. What is the rule for Asterisk to Asterisk-addons compatibility ? Is this rule implicit (any Asterisk-addons 1.6.1.X is compatible with any Asterisk 1.6.1.Y) or did I miss something ? That should be completely correct, unless a security fix requires an API change. I don't think we've ever had that situation, so we've never had to address how we would label the difference. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cache sound files for faster processing
On Tue, Apr 6, 2010 at 12:36 AM, huu giang huugiang...@yahoo.com wrote: Dear List, Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk, it will load from the memory and so Asterisk can process much more call at a time than with faster speed it is not caching. Thanks, Aside from the suggestions, you could try out an SSD drive, which is both expensive compared to a traditional hard drive and very fast. -- _ -- 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] testexpr2
I'm trying to build it and run into all sorts of problems. First, make testexpr2 doesn't work at top level, nor in the main subdirectory. If I manually try the commands for it in main/Makefile, it doesn't have a main and calls ast_log. If use -DSTANDALONE2 instead, those go away, but then: ast_expr2f.o: In function `__register_file_version': ast_expr2f.c:(.text+0xf): undefined reference to `ast_register_file_version' ast_expr2f.o: In function `__unregister_file_version': ast_expr2f.c:(.text+0x1f): undefined reference to `ast_unregister_file_version' ast_expr2f.o: In function `ast_expr': ast_expr2f.c:(.text+0x3e19): undefined reference to `ast_copy_string' Has this been deprecated? -- _ -- 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] RES: Cache sound files for faster processing
Did you tried the good old ram disk? Flavio E. Goncalves www.asteriskguide.com -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de David Backeberg Enviada em: Tuesday, April 06, 2010 12:50 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Cache sound files for faster processing On Tue, Apr 6, 2010 at 12:36 AM, huu giang huugiang...@yahoo.com wrote: Dear List, Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk, it will load from the memory and so Asterisk can process much more call at a time than with faster speed it is not caching. Thanks, Aside from the suggestions, you could try out an SSD drive, which is both expensive compared to a traditional hard drive and very fast. -- _ -- 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] polarity reverse
Is the call successfull? The 'Ignore polarity reversal on line seizure' may just be a warning. What equipment, which Telco is the FXO card connected to? Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas Gulbinskas Sent: Wednesday, 7 April 2010 12:03 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] polarity reverse Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no answeronpolarityswitch=yes I use asterisk 1.6.2 and sangoma a400 fxo ports. Then i try call i get chan_dahdi.c: Ignore possible polarity reversal on line seizure -- _ -- 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] Limit Number Of Simultaneous Outbound SIP Calls
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, April 06, 2010 9:25 AM On Tue, 6 Apr 2010, Deric Page wrote: Is there a way to limit the number of simultaneous outbound SIP calls made by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but that doesn't seem to be working and one of our engineers says that parameter has been depreciated. How about using GROUP() and GROUP_COUNT()? [Deric Page] Thanks, I was able to use these two functions to get us where we needed to go. Deric -- _ -- 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] IAX Call Rejected (was IAX Problem)
On 4/6/2010 10:31 AM, Steve Edwards wrote: On Tue, 6 Apr 2010, bob gailer wrote: I have2 Trixbox Servers. Each has an IAX trunks to the other. One works the other fails: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/526-09eec7c8, IAX2/InterOffice/210,300,tr) in new stack -- Called InterOffice/210 -- Hungup 'IAX2/InterOffice-7578' == Everyone is busy/congested at this time (1:0/0/1) The only difference I am aware of is that one server has a public IP address, the other is behind a NAT. The trunk from the server with the public address works fine. I added nat=yes to the other's peer details - did not help. What should I do? 0) Use a more descriptive subject. 1) Set auth=plaintext (only for the duration of the debug session) and enable iax2 debugging on the CLI. You should see the NEW request and the passwords in plaintext. Here is the calling server side: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 9ms SCall: 06719 DCall: 0 [67.228.218.114:4569] VERSION : 2 CALLED NUMBER : 210 CODEC_PREFS : (ulaw|alaw|gsm) CALLING NUMBER : 526 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Bob's Office LANGUAGE: en FORMAT : 4 CAPABILITY : 14 ADSICPE : 2 DATE TIME : 2010-04-06 14:57:22 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 9ms SCall: 1 DCall: 06719 [67.228.218.114:4569] Here is the called server side: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00019ms SCall: 10453 DCall: 0 [75.177.136.251:4569] VERSION : 2 CALLED NUMBER : 210 CODEC_PREFS : (ulaw|alaw|gsm) CALLING NUMBER : 526 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Bob's Office LANGUAGE: en FORMAT : 4 CAPABILITY : 14 ADSICPE : 2 DATE TIME : 2010-04-06 14:58:02 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 10453 DCall: 1 [75.177.136.251:4569] Verify the username, password, context, and extension all exist. I do not understand or see username. I do not see context. I do not see or have passwords or know how to specify them. Extension exists. I can call the other way with no problem. -- Bob Gailer 919-636-4239 Chapel Hill NC -- _ -- 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] polarity reverse
call not succsessful. I use nokia gsm gw witch have polarity reverse i try on my old asterisk 1.4.17 with digium tdm800 with fxo ports card polarity reverse works fine. But then i connect to asterisk 1.6.2 with sangoma a400 with fxo ports card polarity don't work. polarity reverse is 600 milliseconds set on nokia gsm gw On Apr 6, 2010, at 10:08 PM, Alec Davis wrote: Is the call successfull? The 'Ignore polarity reversal on line seizure' may just be a warning. What equipment, which Telco is the FXO card connected to? Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas Gulbinskas Sent: Wednesday, 7 April 2010 12:03 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] polarity reverse Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no answeronpolarityswitch=yes I use asterisk 1.6.2 and sangoma a400 fxo ports. Then i try call i get chan_dahdi.c: Ignore possible polarity reversal on line seizure -- _ -- 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] IAX Call Rejected (was IAX Problem)
On Tue, 6 Apr 2010, bob gailer wrote: Verify the username, password, context, and extension all exist. I do not understand or see username. I do not see context. I do not see or have passwords or know how to specify them. Extension exists. I can call the other way with no problem. Sorry for the delay, today is a bit busy :) See if this can help: http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication -- 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] testexpr2
On Tuesday 06 April 2010 10:56:56 Richard Kenner wrote: I'm trying to build it and run into all sorts of problems. First, make testexpr2 doesn't work at top level, nor in the main subdirectory. If I manually try the commands for it in main/Makefile, it doesn't have a main and calls ast_log. If use -DSTANDALONE2 instead, those go away, but then: ast_expr2f.o: In function `__register_file_version': ast_expr2f.c:(.text+0xf): undefined reference to `ast_register_file_version' ast_expr2f.o: In function `__unregister_file_version': ast_expr2f.c:(.text+0x1f): undefined reference to `ast_unregister_file_version' ast_expr2f.o: In function `ast_expr': ast_expr2f.c:(.text+0x3e19): undefined reference to `ast_copy_string' Has this been deprecated? Why aren't you using check_expr in the utils directory? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] testexpr2
Why aren't you using check_expr in the utils directory? Aren't they two different things? I thought check_expr looks at a whole file for syntax errors while testexpr2 just parses one expression and returns its value. But if testexpr2 doesn't exist anymore, shouldn't the documentation be updated? -- _ -- 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] dialplan checker
Hello gang, Is there a piece of software out there that can validate a dialplan before I run it though my asterisk (1.4 and 1.6)? Right now I'm just doing live run-time debugging, but that's slow and not always accurate and my dialplan now exceeds 2000 lines. Any ideas? Thanks Danny Nicholas -- _ -- 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] Busy(20) returns non-zero and exits immediately on IAX channel
Hi, I'm running Asterisk 1.4.26.3 and I've noticed an interesting problem when trying to play a Busy tone over a IAX trunk from the PSTN. It seems as though Busy(20) returns non-zero immediately (it does not wait 20s), so the caller never hears the busy tone, but the call just appears to hang up. I don't believe this happens when trying to play a Busy on a SIP trunk. The busy part of the dialplan looks like this, exten = s-BUSY,1,Noop(Dial failed due to trunk reporting BUSY - giving up) exten = s-BUSY,n,Playtones(busy) exten = s-BUSY,n,Busy(20) The only way to remedy this is to put a Wait(20) between the Playtones() and Busy(). Any ideas on why this only fails on IAX and not SIP? Thank you. -- James -- _ -- 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] polarity reverse
Does TDM800 with FXO ports work with 1.6.2? You should have also got other 'polarity related messages' during the call setup. One in particluar which prints debug info when a DAHDI_EVENT_POLARITY get fired. Code below. ast_debug(1, Polarity Reversal event occured - DEBUG 2: channel %d, state %d, pol= %d, aonp= %d, honp= %d, pdelay= %d, tv= %d\n, p-channel, ast-_state, p-polarity, p-answeronpolarityswitch, p-hanguponpolarityswitch, p-polarityonanswerdelay, ast_tvdiff_ms(ast_tvnow(), p-polaritydelaytv) ); If it doesn't work with the TDM800, file a bug report. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas Gulbinskas Sent: Wednesday, 7 April 2010 8:00 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] polarity reverse call not succsessful. I use nokia gsm gw witch have polarity reverse i try on my old asterisk 1.4.17 with digium tdm800 with fxo ports card polarity reverse works fine. But then i connect to asterisk 1.6.2 with sangoma a400 with fxo ports card polarity don't work. polarity reverse is 600 milliseconds set on nokia gsm gw On Apr 6, 2010, at 10:08 PM, Alec Davis wrote: Is the call successfull? The 'Ignore polarity reversal on line seizure' may just be a warning. What equipment, which Telco is the FXO card connected to? Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas Gulbinskas Sent: Wednesday, 7 April 2010 12:03 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] polarity reverse Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no answeronpolarityswitch=yes I use asterisk 1.6.2 and sangoma a400 fxo ports. Then i try call i get chan_dahdi.c: Ignore possible polarity reversal on line seizure -- _ -- 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 -- _ -- 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] RPID on called party
https://issues.asterisk.org/view.php?id=6643 CP On Thu, Apr 1, 2010 at 7:36 AM, Ondrej Valousek webs...@s3group.cz wrote: Hello, Did anyone manage to force asterisk to put Remote-party-ID attribute on the SIP outgoing call? I.e. When A calls B, I want that A gets a name of B displayed on his phone. Note that name of A gets displayed on the B's phone fine, but this is not what I want. This works with Cisco Call manager fine - the RPID is sent as a part of the response to the SIP INVITE this way: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport From: Ondrej Valousek sip:7...@192.168.60.20 sip:7...@192.168.60.20 ;tag=as4786d518 To: sip:1...@192.168.62.12 sip:1...@192.168.62.12 ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104 Date: Tue, 30 Mar 2010 13:53:15 GMT Call-ID: 465a9c200587260d164f451409489...@192.168.60.20 CSeq: 102 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence *Remote-Party-ID: Paul Ryan sip:1...@192.168.62.12 sip:1...@192.168.62.12 ;party=called;screen=yes;privacy=off* Contact: sip:1...@192.168.62.12:5060 sip:1...@192.168.62.12:5060 Content-Length: 0 But I can not make it working with Asterisk. Does anyone have any glue how to achieve this WITHOUT patching asterisk? I am happy to upgrade to the latest/greatest version, I just do not want to patch. Many thanks, Ondrej -- _ -- 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
