[asterisk-users] Quintum Tenor AX and Echo
I have a Quintum AX Tenor gateway sending calls to Asterisk from BT analogue lines connected to FXO. The agents hear an echo on their side but incoming callers hear the conversation fine. I can't seem to find the problem. Anyone seen this issue before? ---DISCLAIMER The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free. -- _ -- 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] Modifying cid.cid_name in app_parkandannounce.c
Am 10.10.10 15:46, schrieb dotnetdub:
> Hi List,
>
> I need to modify the callerID name of the call coming back when a parked
> call returns to the extension that parked it when it times out.
>
> Looking at app_parkandannounce.c
>
> /* Now place the call to the extention */
>
> snprintf(buf, sizeof(buf), "%d", lot);
> memset(&oh, 0, sizeof(oh));
> oh.parent_channel = chan;
> oh.vars = ast_variable_new("_PARKEDAT", buf);
> dchan = __ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR,
> dialstr,3, &outstate, chan->cid.cid_num, chan->cid.cid_name, &oh);
>
> I assume (I hope not incorrectly) that I have to modify the
> variable chan->cid.cid_name
>
>
> Could one of the Asterisk gurus point me in the right direction as to how to
> do this?
>
> Thanks in advance
> Brian
>
>
Hello brian,
this depends on your asterisk version. below 1.8 you would be fine with
chan->cid.cid_name but to be honest use the buildin CALLERID(name)
function would be much easier than changing this by yourself. Or just
have a look how this function sets a calleridname.
with 1.8 or trunk it would be chan->caller.id.name.
there is also a asterisk-dev list to ask specially this questions which
depends on asterisk code itself ;)
best regards
stefan
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Re: [asterisk-users] Quintum Tenor AX and Echo
On 10/11/2010 09:07 AM, Antonio Berrios wrote: >I have a Quintum AX Tenor gateway sending calls to Asterisk from BT > analogue lines connected to FXO. > The agents hear an echo on their side but incoming callers hear the > conversation fine. I can't seem to find the problem. Anyone seen this > issue before? > > > ---DISCLAIMER /> > The information contained in this message is private and confidential and > intended only for the recipient named above. If you are not the intended > recipient you are notified that any communication, circulation or copying of > the information contained in this message is strictly prohibited. If you have > received this message in error please notify us immediately by telephone in > order that we are made aware of this fact and the message can be returned to > us at our address as indicated above. Activity and use of the Sheffield City > Taxis e-mail service is monitored to secure its effective operation and for > other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: > 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. > Sheffield City Taxis Limited uses regularly updated anti-virus software in an > attempt to reduce the possibility of infection. However we do not guarantee > that any attachments to this e-mail are virus free. > > sorry, posted incorrectly. ---DISCLAIMER The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free. -- _ -- 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] Quintum Tenor AX and Echo
Let's try this again. I have a Quintum AX Tenor gateway sending calls to Asterisk from BT analogue lines connected to FXO. The agents hear an echo on their side but incoming callers hear the conversation fine. I can't seem to find the problem. Anyone seen this issue before? ---DISCLAIMER The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free. -- _ -- 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] About Action Originate
I use the action Originate，i want the called first ringing，the called answer,callee ringing.it can achieve? Best regards, justhinker -- _ -- 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] OpenR2
On Mon, Oct 11, 2010 at 01:08:50AM -0300, Flavio Miranda wrote: > > Hi all, > Is it Openr2 supported by asterisk 1.6.2 without pach instalation ? Yes. Provided you have libopenr2. > I am a little bit confuse about that. My asterisk 1.6.2 show me the > following warning: > Unknown signalling method 'mfcr2' at line 29. Most likely you didn't have libopenr2 (or its development headers, depending on the installaation type) > I had downloaded and instaled openr2-1.3.0 but the messages is still shown. > Which files I must to change in order to have everything working properly. A. You need to re-run the configure script of asterisk and rebuild it. B. How have you installaed openr2? From source or from a binary package? If from a binary package, you probably also need the -dev / -devel package. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Setting up realtime config.
On Sun, Oct 10, 2010 at 06:29:27PM -0600, Mike Diehl wrote: > Asterisk replied: > > Peer test not found. > > So it looks like I'm missing something pretty basic. I would suggest to check extconfig.conf. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) -- _ -- 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] About Action Originate
You need to create a dialplan context to achieve it and then access it using originate. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-11 5:54 AM, "施铁泉" wrote: I use the action Originate，i want the called first ringing，the called answer,callee ringing.it can achieve? Best regards, justhinker -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
On 10/09/2010 01:34 PM, bruce bruce wrote: > And that is exactly what is done on the device: Nat=yes but Asterisk > still sees the SIP packet coming in to register with a local IP an so it > responds to a local IP which doesn't even exist on the Asterisk network. > This is what frustrates me that it's not so straight forward to Asterisk > to obtain the proper public IP of the device from the IP packet headers > rather than the SIP packets. 'nat=yes on the device' doesn't really make any sense; does that mean you set some sort of NAT setting on the *device* itself, or does it mean you set 'nat=yes' in the device's peer entry in the Asterisk sip.conf file? If 'nat=yes' is set in the relevant sip.conf peer entry for that device, and Asterisk is properly selecting that entry when the device registers, then Asterisk *will* respond the device's "apparent" IP address and port number, regardless of the address the device includes in the Contact header of the REGISTER request. Setting 'nat=yes' is exactly how you tell Asterisk to use the IP address from the IP header of the packet instead of the address in the SIP message. As I said before, there are likely hundreds of thousands (if not millions) of endpoints registering to Asterisk systems all over the world every day using this mechanism and it works just fine. If it's not working for you, there is some sort of configuration problem. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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
[asterisk-users] Call Failed Audio
Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it's busy or unallocated number or failed with some other reason. Thanks, Deepika -- _ -- 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] iax2 users calls limit for outgoing / incoming
Dear All, I want set call limit for IAX2 users at the time incoming and outgoing, Please help me how i can set call limit as asterisk support for SIP users. -- Thanks & Regards, M. Asif Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenR2
Thanks for while!! I will do that! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda > Date: Mon, 11 Oct 2010 12:06:04 +0200 > From: tzafrir.co...@xorcom.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] OpenR2 > > On Mon, Oct 11, 2010 at 01:08:50AM -0300, Flavio Miranda wrote: > > > > Hi all, > > Is it Openr2 supported by asterisk 1.6.2 without pach instalation ? > > Yes. Provided you have libopenr2. > > > I am a little bit confuse about that. My asterisk 1.6.2 show me the > > following warning: > > Unknown signalling method 'mfcr2' at line 29. > > Most likely you didn't have libopenr2 (or its development headers, > depending on the installaation type) > > > I had downloaded and instaled openr2-1.3.0 but the messages is still shown. > > Which files I must to change in order to have everything working properly. > > A. You need to re-run the configure script of asterisk and rebuild it. > > B. How have you installaed openr2? From source or from a binary package? > If from a binary package, you probably also need the -dev / -devel > package. > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > -- > _ > -- 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] Synchron Playback to caller AND callee ?
Hello, i use the DYNAMIC_FEATURES (features.conf) to start a macro during a call, to start the Monitor application. In this macro i have a Playback to announce the recording. But the Playback play the soundfile only to the caller or to the callee depending if in features.conf the dynamic feature is configured with "self" or "peer" e.g. [applicationmap] ;features.conf automon => *1,peer,Macro,automon [macro-automon] ;extensions.conf exten => s,1,Playback(start-record) exten => s,n,Monitor(wav,myfilename) But i want to have the Playback to both sides, to caller and callee. Is this possible? Any tricks, patch, backports? Kristijan -- _ -- 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 OUtbound IVR Recording
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jayson Baker
Sent: Saturday, October 09, 2010 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk OUtbound IVR Recording
cmd record ?
On Sat, Oct 9, 2010 at 1:28 AM, Govind, Mahesh (NSN - IN/Bangalore)
wrote:
HI,
I have a scenario like the following .
A user clicks on the web page . This triggers an outbound call to users
phone number .
Now the user has to leave a message .
What is the best way of doing this ? Do we have any example of such a
dial plan .
Regards
Mahesh
This is a simple context that plays "static" messages welcome, important and
calllater. It plays a "passed" message as well. To use as you want, just
replace Background(${Data}) with Record(${Data}.gsm). Lines 3-4 incorporate
a wait if the call isn't a SIP line because DAHDI has a 3-7 second delay on
Answer (worse if calling a cell phone).
[accept]
exten => s,1,Answer
exten => s,n,Set(IVRTRY=0)
exten => s,n,Gotoif($["${EXTEN}" > "SIP"]?start)
exten => s,n,Wait(9)
exten => s,n(start),Background(welcome)
exten => s,n,Background(important)
exten => s,n,WaitExten(5,m)
exten => s,n,Set(IVRTRY=$[${IVRTRY} +1])
exten => s,n,Verbose(Try ${IVRTRY})
exten => s,n,Gotoif($["${IVRTRY}" < "4"]?accept|s|start)
exten => s,n,Goto(end-call|s|1)
exten => 1,1,ForkCDR(v,s(fullcmd=${Data}))
exten => 1,n,Background(${Data})
exten => 1,n,Background(repeatmsg)
exten => 1,n,WaitExten(5,m)
exten => 1,n,Goto(end-call|s|1)
exten => 2,1,Background(calllater)
exten => 2,n,ForkCDR(v,s(reject=${Data}))
exten => 2,n,Goto(end-call|s|1)
exten => 3,1,Goto(accept|1|2)
exten => *,1,Goto(accept|s|1)
exten => i,1,Goto(accept|s|1)
exten => t,1,Goto(accept|s|1)
--
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Re: [asterisk-users] iax2 users calls limit for outgoing / incoming
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mian Asif Sent: Monday, October 11, 2010 7:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] iax2 users calls limit for outgoing / incoming Dear All, I want set call limit for IAX2 users at the time incoming and outgoing, Please help me how i can set call limit as asterisk support for SIP users. I haven't done this myself, but I read a lot about the Group() function and how it can be used for things like this. -- _ -- 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] Create channel bank with TDMoE
Hello, I want to create channel bank in this case: "channel bank" |-| | FXS,FXO<->TDMoE<--|-->Asterisk |-| How can it? -- _ -- 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] Create channel bank with TDMoE
On Mon, Oct 11, 2010 at 06:18:24PM +0330, Karim Davoodi wrote: > Hello, > I want to create channel bank in this case: > >"channel bank" > |-| > | FXS,FXO<->TDMoE<--|-->Asterisk > |-| > > How can it? Buy one? Build one? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Create channel bank with TDMoE
Hi, Here in Brazil we've got a company named CIANET. They do exactly you want. I am engineer and work on it with them. http://www.cianet.ind.br/pt/channel_bank.php Thank you Luis A P Barbosa 2010/10/11 Karim Davoodi > Hello, > I want to create channel bank in this case: > > "channel bank" > |-| > | FXS,FXO<->TDMoE<--|-->Asterisk > |-| > > How can it? > > -- > _ > -- 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] Create channel bank with TDMoE
Karim Davoodi wrote: > Hello, > I want to create channel bank in this case: > >"channel bank" > |-| > | FXS,FXO<->TDMoE<--|-->Asterisk > |-| > > How can it? > http://www.voipon.co.uk/redfone-fonebridge2-quad-t1e1-ethernet-bridge-p-348.html -- _ -- 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] Call Failed Audio
Sorry this is a list for the Asterisk GUI Project. I think you may have better luck on the FreePBX list / forums. ~ Andrew "lathama" Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Mon, Oct 11, 2010 at 9:33 AM, Deepika Nijhawan wrote: > Hi, > > > > On freepbx (GUI), whatever reason number fails we always get 'all circuits > are busy' audio. > > Does anybody know how to get far end audio when we dial wrong number or when > it’s busy or unallocated number or failed with some other reason. > > > > > > Thanks, > > Deepika > > > > -- > _ > -- 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] Setting up realtime config.
On Monday 11 October 2010 4:34:46 am Stefan Tichy wrote: > On Sun, Oct 10, 2010 at 06:29:27PM -0600, Mike Diehl wrote: > > Asterisk replied: > > > > Peer test not found. > > > > So it looks like I'm missing something pretty basic. > > I would suggest to check extconfig.conf. That's where the problem was; I had the wrong section title. Thanks, all. -- Take care and have fun, Mike Diehl. -- _ -- 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] Unable to find a codec translation path from ulaw|h261 to slin
Never mind...
I mistakenly interpreted codec_a_mu.so as some sort of universal translator
between ulaw, alaw, and slin. When I loaded the "rest" of the modules, it
worked like a champ.
Mike.
On Sunday 10 October 2010 6:57:30 pm Mike Diehl wrote:
> I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and
> I've encountered a problem playing back a .wav file to an Ekiga client:
>
> My dialplan looks like:
>
> exten => 730,1,answer
> exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign)
> exten => 730,n,hangup
>
> Sovereign.wav is a .wav file that plays nicely on my 1.4 server.
>
> Here is what the console displays:
>
> -- Executing [...@customers:2] Playback("SIP/user_xxx-0012",
> "/home/phones/common/moh/moha/Sovereign") in new stack
> Unable to find a codec translation path from 0x40004 (ulaw|h261) to 0x40
> (slin)
> Unable to open /home/phones/common/moh/moha/Sovereign (format 0x40004
> (ulaw| h261)): No such file or directory ast_streamfile failed on
> SIP/user_xxx-0012 for /home/phones/common/moh/moha/Sovereign
>
> I was under the impression that I didn't have to do anything to get slin
> support.
>
> when I do: module show like codec_
> I get:
>
> Module Description Use
> codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0
> codec_gsm.so GSM Coder/Decoder0
> codec_ulaw.so mu-Law Coder/Decoder 0
>
> I'm assuming use=0 because the server is idle.
>
> I've got allow = all in my sip.conf file.
>
> Anyway, does anyone have an idea on how to resolve this?
--
Take care and have fun,
Mike Diehl.
--
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Re: [asterisk-users] TDM 400p and Noise on the line
> Hi > > I wonder if anyone has any sugestions > > > I have had a TDM400 for a couple of years, and I have always had problems > with noise on the line, so tonight I have been doing some research and have > found that if I load the CPU dahdi_test has almost perfect results and no > noise > > dahdi_test > Opened pseudo dahdi interface, measuring accuracy... > 99.997% 99.999% 99.998% 99.997% 99.999% 99.998% 99.998% 99.998% > 99.998% 99.998% 99.998% 99.997% 99.998% 99.997% 99.998% 99.998% > 99.998% 99.998% 99.998% 99.997% 99.999% 99.998% 99.998% 99.997% > 99.998% > --- Results after 25 passes --- > Best: 99.999 -- Worst: 99.997 -- Average: 99.997895, Difference: 100.002028 > > > > but when the CPU is not loaded there is white noise low volume and the > results of dahdi_test > > dahdi_test > Opened pseudo dahdi interface, measuring accuracy... > 99.992% 99.988% 99.951% 99.991% 99.992% 99.992% 99.992% 99.992% > 99.991% 99.991% 99.992% 99.991% 99.991% 99.952% 99.992% 99.992% > 99.992% 99.992% 99.951% 99.992% 99.992% 99.992% 99.992% 99.950% > 99.991% 99.991% 99.991% 99.992% 99.992% 99.951% 99.992% 99.992% > 99.992% 99.991% 99.952% 99.992% 99.992% 99.991% 99.992% 99.951% > 99.991% > --- Results after 41 passes --- > Best: 99.992 -- Worst: 99.950 -- Average: 99.984461, Difference: 100.001168 > > > Could you explain how I can improve the call quality without running the > CPU at 99% al the time? > > running the latest Dahdi 2.4.0 Echo Canceller: OSLEC and asterisk 1.6.2.13 I suspect that you might be seeing some effect of the CPU going into, and out of an IDLE state... possibly due to the use of the HLT instruction in the kernel's idle loop, or possibly due to the ACPI BIOS (or the operating system "cpufreq" support code) changing the CPU clock rate or voltage. These sorts of changes in processor state might have several sorts of effects on the TDM400P card and the audio it is processing: - Changes in processor state (e.g. core voltage or clock speed) can cause a brief interrupting in processing... the CPU instruction processing must sometimes be halted in order to allow the clock PLL to re-lock at a different rate and for the core voltage to stabilize. This might possibly be adding enough latency to interrupt service time to affect the card (e.g. losing some audio samples), or skewing the timing enough that OSLEC's echo cancelling algorithms exhibit different behavior. - Changes in the amount of power being drawn by the CPU, when it goes from flat-out processing to idle, can be quite substantial in modern CPUs. Some of today's multi-core CPUs dissipate on the order of 100 watts during full-speed processing but drop down far below that when idle. The amount of current being drawn by the processor will cause changes in the voltage on the motherboard's +12 and +5 supply lines, and these voltage changes are likely to reach the TDM400P. If the TDM400P doesn't have good on-board voltage regulation and noise filtering (and I suspect that it might not), then some of the voltage noise on the supply rails could leak into the audio. Similar problems exist with many PC on-the-motherboard audio interfaces. As to how to fix it? Well, you're going to have to experiment. The first thing I'd suggest trying, would be to see if you can disable any sort of ACPI- or kernel-based "power saving" adjustment of the CPU's clock speed and core voltage. This might involve disabling SpeedStep (or the equivalent) in the BIOS, or switching Linux from using the "ondemand" power governor to a single-speed one (either "performance" or "powersave" or "usermode"). Possibly, running Asterisk at a real-time priority might reduce the issue, if it's timing-related. If it's simply a matter of noise on the power rails, you may not be able to get rid of it at all easily... might have to change motherboards, power supplies, or switch to an external phone interface device which is inherently immune to electrical noise within the PC chassis. -- _ -- 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] don't leave meetme conf if key pressed
Hi @ all, what is the best way to to use features like MeetmeCount without leaving the conference. I use Meetme(,X) and MEETME_EXIT_CONTEXT=context, but the problem is that the caller leave the Conference :( Is it possible to press a key, without this obstacle? Thanx for your answers Daniel Knoll -- _ -- 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] Second time Parking issue
Hello All, I have a issue with park and pickup feature. I have asterisk 1.4.35 branch, Here is the scenario for the park and pickup. I have changed parking feature with *72 for my asterisk in features.conf. When i have inbound call it comes to one extension or ring group and then I put that inbound call in parking. After that I will dial some extension to get that call back, I am using parkedcalls application for that. It is working fine, the issue is now when i have picked up call and connected again then I want to park that inbound call second time, But asterisk is not allowing that dtmf or parking. I am not able to do park same call second time. can any one help me for second time parking on inbound call. Thanks in advance. Thanks, Max Alex -- _ -- 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] Second time Parking issue
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of max.asterisk Sent: Monday, October 11, 2010 3:21 PM To: Asterisk Developers Mailing List; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Second time Parking issue Hello All, I have a issue with park and pickup feature. I have asterisk 1.4.35 branch, Here is the scenario for the park and pickup. I have changed parking feature with *72 for my asterisk in features.conf. When i have inbound call it comes to one extension or ring group and then I put that inbound call in parking. After that I will dial some extension to get that call back, I am using parkedcalls application for that. It is working fine, the issue is now when i have picked up call and connected again then I want to park that inbound call second time, But asterisk is not allowing that dtmf or parking. I am not able to do park same call second time. can any one help me for second time parking on inbound call. Thanks in advance. Thanks, Max Alex What does your features.conf look like? Have you looked at the CLI output during this process (or read /var/log/asterisk/full)? -- _ -- 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] user number in conference
Hey, i forgot to ask, how can i get the user number from a caller he is in a conference, i don't find a variable to us this for the current channel. Only the command "meetme list " shows the usernumber, but i can't use this output. Thanks. Daniel -- _ -- 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] MWI Assistance
Hi, I'm struggling to get the MWI set up on a few Polycom phones. The setup is like this. I've got a few phones in the context called [company2_phones] and I've got a few mailboxes in the voicemail context [company2]. Therefore, for each entry in sip.conf (i'm actually using sip realtime if that makes a difference), i've entered "mailbo...@company2" (1 being the name of the mailbox) However, the phone doesnt subscribe to the mailbox status. In the Polycom documentation, it asks me to provide:- ASCII encoded string containing digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL (6416 or 6...@polycom.com) But I have no idea what to enter. I've tried everything I can think of but I get this in the Asterisk CLI:- [2010-10-11 23:06:08] NOTICE[18424]: chan_sip.c:16331 handle_request_subscribe: Received SIP subscribe for peer without mailbox: company2_201 company2_201 is the user part listed in sip.conf for that particular extension. What do I enter in order to get it to request the mailbox status? Any assistance would be appreciated. Thanks Dan -- _ -- 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] user number in conference
On Tue, 12 Oct 2010, Daniel Knoll wrote: > i forgot to ask, how can i get the user number from a caller he is in a > conference, i don't find a variable to us this for the current channel. > Only the command "meetme list " shows the usernumber, but i > can't use this output. If you use AMI in an AGI you can parse this output and do something useful with it like mute the user or return the user number to your dial plan as a channel variable -- 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] MWI Assistance
> I'm struggling to get the MWI set up on a few Polycom phones. Sorted. From voip-info. http://www.voip-info.org/wiki/view/Asterisk+RealTime The database peers/users are not kept in memory. These are only loaded when we have a call and then deleted, so there's no support for NAT keep-alives (qualify=) or voicemail indications for these peers. NOTE: If you enable RealTime caching in your sip.conf, Voicemail MWI works and so does 'sip show peers' - see rtcachefriends=yes. The downside to this is that if you change anything in the database, you need to do a 'sip reload' (for major changes) or 'sip prune realtime PEERNAME' (for single peer changes) before they become active. -- _ -- 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 and ANI
Hi All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ANI to whatever and CID Number to a different whatever so on the other end of the PRI you will receive the two different values? Is this correct or is there a way to set ANI on an outgoing SIP channel (like to a PRI gateway) and the gateway will see a CID Number and a separate ANI and insert that into the ISDN messaging down the PRI? Thanks for any clarification. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Receive Call from unknown user
Hello List,
I have noticed for the past few weeks that someone from an unknown IP is
trying to make a call to my Asterisk box, below is the sample content of the
log file. Sometimes the calls are being made every seconds.
Is my system being hack by someone?
Oct 12 09:41:47] VERBOSE[3114] netsock.c: == Using SIP VRTP CoS mark 6
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing
[99541442073479...@from-sip-external:1] NoOp("SIP/113.105.153.251-0265",
"Received incoming SIP connection from unknown peer to 9954144207347")
in new stack
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing
[99541442073479...@from-sip-external:2] Set("SIP/113.105.153.251-0265",
"DID=9954144207347") in new stack
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing
[99541442073479...@from-sip-external:3] Goto("SIP/113.105.153.251-0265",
"s,1") in new stack
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Goto (from-sip-external,s,1)
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing [...@from-sip-external:1]
GotoIf("SIP/113.105.153.251-0265", "0?checklang:noanonymous") in new
stack
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Goto (from-sip-external,s,5)
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing [...@from-sip-external:5]
Set("SIP/113.105.153.251-0265", "TIMEOUT(absolute)=15") in new stack
[Oct 12 09:41:47] VERBOSE[21627] func_timeout.c: Channel will hangup at
2010-10-12 09:42:02.042 PHT.
[Oct 12 09:41:47] VERBOSE[21627] pbx.c: -- Executing [...@from-sip-external:6]
Answer("SIP/113.105.153.251-0265", "") in new stack
Please Advise.
Malvin
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