Re: [asterisk-users] Use of 603 Declined

2010-01-29 Thread Olle E. Johansson
Agree that the 603 is wrong. It hasn't caused me issues but I see where it could. And it goes against what I have been teaching in my classes, which is irritating ;-) In Asterisk, it's only used when we have no other hangup cause - and is propably an indication that there is a code path that

Re: [asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short

2010-01-29 Thread Olle E. Johansson
Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 21:41 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short I would very much

Re: [asterisk-users] Use of 603 Declined

2010-01-29 Thread Alex Balashov
I don't know about 4xx, but 503 would be more benign for general/ miscellaneous errors than 603. -- Sent from mobile device On Jan 29, 2010, at 3:54 AM, Olle E. Johansson o...@edvina.net wrote: Agree that the 603 is wrong. It hasn't caused me issues but I see where it could. And it goes

Re: [asterisk-users] Use of 603 Declined

2010-01-29 Thread Olle E. Johansson
29 jan 2010 kl. 10.25 skrev Alex Balashov: I don't know about 4xx, but 503 would be more benign for general/ miscellaneous errors than 603. 503 indicates that there's a problem with the server, so that's not a good replacement. We're sending this when there's a failed call, in most cases

[asterisk-users] chan_mobile problem with audio (distorted)

2010-01-29 Thread Marian Zahariev
Hello to all. I have installed asterisk-1.6.2.1 + asterisk-addons-1.6.2.0 (for chan_mobile) + bluez-4.60. Bluetooth Dongle: Canyon CN-BTU4 (0a12:0001 Cambridge Silicon Radio, Ltd Bluetooth Dongle (HCI mode)) Device Descriptor:

[asterisk-users] VUC Today at 1 PM EST: Counterpath/Bria

2010-01-29 Thread Randy R
Hi, In the aftermath of Digium's and Counterpath's Bria for Asterisk announcement, we're happy to chat with Todd Carothers, Counterpath Product Manager today at 1 PM EST. For more info, http://vuc.me Join us on IRC #vuc on Freenode.net or use the web client at http://vuc.me/irc Call in

[asterisk-users] Address family not supported by protocol

2010-01-29 Thread Chris Gentle
After an upgrade to asterisk 1.6.2.1 I'm unable to make outgoing calls via Vitelity. I get lots of these on my asterisk console: [Jan 27 08:58:41] WARNING[25653]: chan_sip.c:3581 __sip_xmit: sip_xmit of 0x834ae08 (len 927) to 64.2.142.18:0 returned -1: Address family not supported by

[asterisk-users] disable comfort noise

2010-01-29 Thread Szasz Szabolcs
Hi, How can I disable comfort noise on Asterisk? Szabolcs Szasz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] disable comfort noise

2010-01-29 Thread Kevin P. Fleming
Szasz Szabolcs wrote: How can I disable comfort noise on Asterisk? Asterisk does not have a comfort noise generator, so there is nothing to disable. You'll have to be more specific about what you are trying to accomplish. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies

[asterisk-users] 1 Asterisk server, multiple registrations to ITSP

2010-01-29 Thread jonas kellens
Hello list ! Having troubles with multiple registrations to one and the same ITSP. This sip.conf : register = user1:pass...@sip.itsp register = user2:pass...@sip.itsp ; outgoing conversations [user1-out] type=peer host=sip.ITSP username=user1 secret=secret1 fromuser=user1 dtmfmode=rfc2833

Re: [asterisk-users] TDM2400 card FXS problems

2010-01-29 Thread Noah I. Engelberth
I don't know what it showed at that time, I didn't know to look for that. I'll try to get the customer's permission to re-create the symptoms this weekend and post back with the lsdahdi output. Thank you, Noah Engelberth Direct Link Computer Systems - Message: 12 Date: Thu, 28 Jan 2010

Re: [asterisk-users] Cell Phone dialing

2010-01-29 Thread Danny Nicholas
Apparently it is a telco issue. As I stated, when I do the dial from my Polycom 501 handset, the landline call indicates ringing at 0:01 or 0:02 on the timer; the cell call doesn't indicate ringing until 0:04 or 0:05. Manual calls excluding Asterisk from the process produced suitably similar

Re: [asterisk-users] Use of 603 Declined

2010-01-29 Thread Fred Posner
On Jan 29, 2010, at 5:59 AM, Olle E. Johansson wrote: 29 jan 2010 kl. 10.25 skrev Alex Balashov: I don't know about 4xx, but 503 would be more benign for general/ miscellaneous errors than 603. 503 indicates that there's a problem with the server, so that's not a good replacement.

Re: [asterisk-users] TDM2400 card FXS problems

2010-01-29 Thread Allway
Hi Noah, IMHO replace the FXS module with new one or working module. That could be proved where is the root cause. Mostly, the trouble issue is caused by FXS modules in my experience. Good Luck, Johnson On Fri, Jan 29, 2010 at 10:10 PM, Noah I. Engelberth n...@directlinkcomputers.com wrote:

Re: [asterisk-users] disable comfort noise

2010-01-29 Thread Cary Fitch
Something else that is flaky, missing or otherwise irritatingly broken in the piece of shit that is 'Asterisk'. [Cary Fitch] It is an open source project. When can we count on your contribution of a comfort noise generator that will not be a piece of s--t? Can you have that by Monday? CF

Re: [asterisk-users] disable comfort noise

2010-01-29 Thread listu...@spamomania.co.uk
On Fri, 2010-01-29 at 07:29 -0600, Kevin P. Fleming wrote: Szasz Szabolcs wrote: How can I disable comfort noise on Asterisk? Asterisk does not have a comfort noise generator, so there is nothing to disable. You'll have to be more specific about what you are trying to accomplish. --

Re: [asterisk-users] disable comfort noise

2010-01-29 Thread Danny Nicholas
Just read RFC 3389; Guess the solution is going to be to sell Asterisk to someone, make us pay for it and make everyone run a g.711 codec? If you don't like it, don't use it! -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] disable comfort noise

2010-01-29 Thread Steve Howes
On 29 Jan 2010, at 14:53, listu...@spamomania.co.uk wrote: Something else that is flaky, missing or otherwise irritatingly broken in the piece of shit that is 'Asterisk'. It's open source. Fix it yourself, no one else is going to fix it for you with that attitude. --

Re: [asterisk-users] Use of 603 Declined

2010-01-29 Thread Kristian Kielhofner
On Fri, Jan 29, 2010 at 5:59 AM, Olle E. Johansson o...@edvina.net wrote: 29 jan 2010 kl. 10.25 skrev Alex Balashov: I don't know about 4xx, but 503 would be more benign for general/ miscellaneous errors than 603. 503 indicates that there's a problem with the server, so that's not a good

[asterisk-users] Cell phone redialer?

2010-01-29 Thread Myles Wakeham
I have an Asterisk 1.4.2 system installed at our office, and have a few 'on the road' sales people that want to make calls from their cell phones in transit, but they are complaining that people returning calls that they make from their cell phones are simply just using the CID that is coming

Re: [asterisk-users] Cell phone redialer?

2010-01-29 Thread Jeremy Kister
On 1/29/2010 10:13 AM, Myles Wakeham wrote: Basically I wanted to see if I could get them to call our phone number on Asterisk, enter some special extension and/or enter a passcode, and then enter the phone number that they wanted to call in which our phone system would route the call to

Re: [asterisk-users] Cell phone redialer?

2010-01-29 Thread Danny Nicholas
Try this - exten = 393,1,noop(forward this call) - exten = 393,n,authenticate(7277,a) - exten = 393,n,Read(custno,customerphone,11,skip,5,1) - exten = 393,n,Dial(DAHDI/1,w${custno},30,mKkg) - exten = 393,n,Playback(vm-goodbye) - exten = 393,n,Hangup This will require the caller to enter a

[asterisk-users] Problem with ringing (or absence of...) with Polycom forwarding

2010-01-29 Thread Mike
Hi, I`m having a problem I cannot explain. When dialing 555-555- (for example), I get a ringing sound until the person answers. When I have my Polycom forwarded to 555-555-, I do not get the ringing, but it dials fine and eventually when the person answers everything works fine.

Re: [asterisk-users] TDM2400 card FXS problems

2010-01-29 Thread garry liu
Hello Noah, Just shifting your TDM2400 and system to other computer, it might be figured out your issue. As my opinion, the issue is involved with computer reset and TDM2400 epld programming. Would figuring out the issue completely, it is better to replace the TDM2400 card with Digium's. Garry

Re: [asterisk-users] Use of 603 Declined

2010-01-29 Thread Kevin P. Fleming
Kristian Kielhofner wrote: I don't want to ruin your plans for tonight (RFC3261 is a lot of fun) but how about 403: 21.4.4 403 Forbidden The server understood the request, but is refusing to fulfill it. Authorization will not help, and the request SHOULD NOT be repeated. Well,

Re: [asterisk-users] Cell phone redialer?

2010-01-29 Thread Danny Nicholas
Here are snippets from my OP that I tested and the DISA that Jeremy suggested; ; authenticate and dial exten = 393,1,noop(forward this call) exten = 393,n,authenticate(7277,a) exten = 393,n,Read(custno,enter-phone-number10,7,skip,5,1) exten = 393,n,Dial(DAHDI/1/w${custno},30,mKkg) exten =

Re: [asterisk-users] Problem with ringing (or absence of...) withPolycom forwarding

2010-01-29 Thread Danny Nicholas
Please post CLI output from the 2 calls with the number xxx'ed out. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, January 29, 2010 9:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Re: [asterisk-users] Use of 603 Declined

2010-01-29 Thread Kristian Kielhofner
On Fri, Jan 29, 2010 at 10:31 AM, Kevin P. Fleming kpflem...@digium.com wrote: Well, that's the problem, and it's the reason why 603 is so commonly used. This is a situation where the current request has failed, but there is no indication that repeating the request will also fail. 403 means

Re: [asterisk-users] 1 Asterisk server, multiple registrations to ITSP

2010-01-29 Thread Robert Lister
On Fri, 2010-01-29 at 15:09 +0100, jonas kellens wrote: Hello list ! Having troubles with multiple registrations to one and the same ITSP. This sip.conf : register = user1:pass...@sip.itsp register = user2:pass...@sip.itsp ; outgoing conversations [user1-out] type=peer

Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?

2010-01-29 Thread khalid touati
Hi William, I appreciate your answer, though can you make things more clear for me: 1- i am not using extensions when registering PBX boxes in IAX files. 2- is inbounx context in the call sender PBX (pbx1) and outbound context is in the call receiver (or dialer) PBX (pbx2)? 3- i am using two

[asterisk-users] smsq command

2010-01-29 Thread Jerry Geis
Is it possible to use the smsq command in asterisk to send SMS messages to a aggregator. So If I have an IP address, password and port for my connection can I use smsq to send SMS messages? I dont see how to set that up? I am looking: http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms Thanks,

[asterisk-users] Questions about asterisk and spa2102

2010-01-29 Thread Kosa
Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a normal phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to

Re: [asterisk-users] 1 Asterisk server, multiple registrations to ITSP

2010-01-29 Thread jonas kellens
When setting type=friend for the incoming calls : ; outgoing conversations [user1-out] type=peer host=sip.ITSP username=user1 secret=secret1 fromuser=user1 ; incoming conversations [user1] type=friend host=sip.ITSP context=user1incoming ; outgoing conversations [user2-out] type=peer

Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?

2010-01-29 Thread William Stillwell (Lists)
You are using contexts.. Look @ destination pbx, you should see something like this: Rejected connect attempt from ip of source pbx, request 'ext@incoming conext' does not exist If you didn't put a context under the peer, it uses the default one in the iax.conf file which is normally

[asterisk-users] microphone on Polycom 550/650

2010-01-29 Thread hin lee
I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem?

Re: [asterisk-users] disable comfort noise

2010-01-29 Thread adamk
To get back to the original poster's possible situation, i've seen this with my first IP phone, which was a cisco 7912 (SIP image). With that phone, asterisk sometimes gave me this same error. I'm quite sure i've asked the very same question here back then (probably i was a bit more specific

Re: [asterisk-users] Help configuring Audiocodes MP-104 FXO

2010-01-29 Thread Daniel - Asterisk
Just if it is helps someone, based on information at the blog: http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.htmlI've summarized the following steps: *Step 1:* Configure audiocodes to have registration account with asterisk, this can be done easily with Protocol

Re: [asterisk-users] Help configuring Audiocodes MP-104 FXO

2010-01-29 Thread Matt Collins
Damn, where were you 6 months ago? ;) Daniel - Asterisk wrote: Just if it is helps someone, based on information at the blog: http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.html I've summarized the following steps: *Step 1:* Configure audiocodes to have

Re: [asterisk-users] microphone on Polycom 550/650

2010-01-29 Thread Danny Nicholas
You don't state this, but the assumption would be that your external calls are DAHDI based, so you might need to tweak txgain in dahdi.conf. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee Sent: Friday, January 29,

Re: [asterisk-users] 911, location

2010-01-29 Thread Leif Neland
Den 28-01-2010 20:15, Danny Nicholas skrev: Here's one solution: - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4,Background(emergencyin${IMAT}) Where you would record /var/lib/asterisk/sound/emergencyin100

Re: [asterisk-users] TDM2400 card FXS problems

2010-01-29 Thread wassim darwich
HI: I had this problem before with TDM2400P but with fxo modules and VPMADT032 (echo canceller),there was no audio at all.but then i unpulgged the ehco canceller module (VPMADT032) from the TDM2400P board and started the server  and then  i didnt face this issue any more. In your case  first

Re: [asterisk-users] 911, location

2010-01-29 Thread Danny Nicholas
This might help - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4(keepup),Background(emergencyin${IMAT}) - exten = _911,5,wait(10) - exten = _911,6,Goto(keepup) This would repeat the message every 10 seconds... --

[asterisk-users] New feature: Asterisk Manager Interface commands for DeviceState

2010-01-29 Thread Håkon Nessjøen
Hi, I've uploaded a new patch at https://issues.asterisk.org/view.php?id=16732which adds two new AMI commands, called DeviceStateSet and DeviceStateGet. These commands let you update Custom device states, and read all devicestates from AMI. It would be very nice if someone could help me test

Re: [asterisk-users] Help configuring Audiocodes MP-104 FXO

2010-01-29 Thread Daniel - Asterisk
It was a pending draft I forgot to send.. sorry. On Fri, Jan 29, 2010 at 1:23 PM, Matt Collins mcoll...@ccdservice.netwrote: Damn, where were you 6 months ago? ;) Daniel - Asterisk wrote: Just if it is helps someone, based on information at the blog:

[asterisk-users] Digium fax - sending fax call file vs manager originate

2010-01-29 Thread Hristo Benev
Hello, I have Asterisk 1.6.1.12 with FAX For Asterisk Components: Applications: 1.6.1.5_1.1.6 Digium FAX Driver: 1.6.1.5_1.1.6 (optimized for core2_32) If I use call file with spool Channel: SIP/IP/DEst No MaxRetries: 0

[asterisk-users] Broker lines on a T1 : Signaling convention?

2010-01-29 Thread Martin Andrews
I've been running Asterisk with a standard PRI for regular telecoms. This is also connected to our Nortel PBX for 'ordinary users'. The system has been working nicely (including Cisco 7970 phones that are connecting via SIP). But now I'm going 'on net' with broker lines (for a trading room

Re: [asterisk-users] 911, location

2010-01-29 Thread Leif Neland
Den 29-01-2010 19:38, Danny Nicholas skrev: This might help - exten = _911,1,Set(IMAT=EXTEN) - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2) - exten = _911,3,Dial(DAHDI/1,w911) - exten = _911,4(keepup),Background(emergencyin${IMAT}) - exten = _911,5,wait(10) - exten =

Re: [asterisk-users] 911, location

2010-01-29 Thread Danny Nicholas
The idea behind the OP was that the caller was a man down who couldn't speak to 911, just dial the number. You could always change wait to waitexten and make an exten to break the loop if you were able to talk. -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short

2010-01-29 Thread wassim darwich
Hi: i did set the rtp ports in  rtp.conf  to rtpstart=5000 , rtpend=31000 ,and i used canreinvite=no and the problem still exists ,however i did the rtp debug and here is the output :   = Spawn extension (direct, 9613070741, 2) exited non-zero on 'SIP/03070741-083b9da0'     -- Executing

Re: [asterisk-users] 911, location

2010-01-29 Thread Kevin P. Fleming
Leif Neland wrote: 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message.

[asterisk-users] Help for MOH - sounding scratchy/static on hold

2010-01-29 Thread das sandesh
Hi All, I tried using some music on hold (music) files, when I test it with normal SIP phone its clear and good, but when I call from my cell phone or POTS line it sounds a bit scratchy/static and not clear at all, is there any software that i need to install in the asterisk system to make this

Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold

2010-01-29 Thread Danny Nicholas
Mpg123 works well for us. You have to get your files into mp3 format, but LAME does this simply. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of das sandesh Sent: Friday, January 29, 2010 4:26 PM To: Asterisk Users Mailing

Re: [asterisk-users] Unable to create channel of type 'DAHDI'(cause 0- Unknown)

2010-01-29 Thread sean darcy
listu...@spamomania.co.uk wrote: On Thu, 2010-01-28 at 23:11 -0600, Karl Fife wrote: Appears completely resolved! No more home-spun patches! Thanks! -K It's *not* fixed here: DAHDI Version: 2.2.1 Echo Canceller: MG2 But as is depressingly the 'norm' for Asterisk it comes back to

Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold

2010-01-29 Thread Jeff Brower
Sandesh- I tried using some music on hold (music) files, when I test it with normal SIP phone its clear and good, but when I call from my cell phone or POTS line it sounds a bit scratchy/static and not clear at all, is there any software that i need to install in the asterisk system to make

[asterisk-users] callerid not working over sip

2010-01-29 Thread sean darcy
Calling from my home using Asterisk 1.6.2.1 to an office extension (Asterisk 1.6.1.13) the callerid is not honored: Home: -- Starting simple switch on 'DAHDI/1-1' -- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@internal:2] NoOp(DAHDI/1-1, Context:

[asterisk-users] Caller ID not working properly on some phones...

2010-01-29 Thread Carlos Chavez
I have a strange problem with CallerID that only affects some phones. The problem is that whenever I receive a call the Callerid Name is correct but the Callerid number is always my own extension. It does not matter if the call is internal or external. So far only Aastra phones and

Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold

2010-01-29 Thread Steve Edwards
On Fri, 29 Jan 2010, Danny Nicholas wrote: Mpg123 works well for us. You have to get your files into mp3 format, but LAME does this simply. Why would you want to compress files when you will have to decompress them again every single time the are used? I'd rather use the CPU cycles to

[asterisk-users] Asterisk status 488 Not acceptable here on receiving fax

2010-01-29 Thread Deepesh D
Hello, I have been trying to setup asterisk 1.6.1.1 to receive fax. Whenever a SIP peer (zoiper soft phones) tries to send a fax message asterisk responds by sending a 488 Not acceptable here and the sending fails. I tried changing a few sip settings like canreinvite and codec preferences, but it

Re: [asterisk-users] Unable to create channel of type 'DAHDI'(cause 0- Unknown)

2010-01-29 Thread Matt Riddell
On 30/01/10 11:48 AM, sean darcy wrote: Sigh. OK you don't like asterisk - sorry. Obviously some other software works better for you. I'm glad. Don't worry, he/she's trolling, second post like that for the day :) Obviously has an issue with something, but rather than try and get it sorted

Re: [asterisk-users] Questions about asterisk and spa2102

2010-01-29 Thread John Novack
Kosa wrote: Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a normal phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline.

Re: [asterisk-users] Questions about asterisk and spa2102

2010-01-29 Thread Steve Edwards
On Fri, 29 Jan 2010, Kosa wrote: 1.- is it possible to use an spa2102 to make and revice calls from a normal phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. The 2102 is an FXS

[asterisk-users] Astribank problem

2010-01-29 Thread frangky robert
H all... I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final. My problem is, every time i unplug the astribank power supply, and reconnect it, astribank cannot work again (lsusb result is 11x0)... but, after reinstall

Re: [asterisk-users] disable comfort noise

2010-01-29 Thread uzzi
On Fri, Jan 29, 2010 at 1:14 PM, ad...@3a.hu wrote: To get back to the original poster's possible situation, i've seen this with my first IP phone, which was a cisco 7912 (SIP image). With that phone, asterisk sometimes gave me this same error. I'm quite sure i've asked the very same