[asterisk-users] PITCH_SHIFT()

2011-12-19 Thread John Jolly
This list is a great resource and I thank all the Asterisk Guru's who actively contribute to it. In Leif Madsen's AstriCon 2010 talk titled "5 Things You Didn't Know Asterisk Could Do

Re: [asterisk-users] Use different local IP for each SIP trunk

2011-12-19 Thread José Pablo Méndez Soto
May I ask why do you need different IP addresses to source calls? I mean, its not a common practice, would like to understand the idea behind it. *José Pablo Méndez * On Mon, Dec 19, 2011 at 11:07 PM, Anton Kvashenkin wrote: > AFAIK you can add exterin= in sip.conf for each trunk, corr

Re: [asterisk-users] Sending Fax Dialplan with Retry Attempt

2011-12-19 Thread Zoel Hairi
> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Zoel Hairi > Sent: Monday, December 19, 2011 11:21 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Sendin

Re: [asterisk-users] Use different local IP for each SIP trunk

2011-12-19 Thread Anton Kvashenkin
AFAIK you can add exterin= in sip.conf for each trunk, correct me if i'm wrong. 2011/12/20 Douglas Mortensen > Hello, > > ** ** > > I have a SIP provider whom I may want to have multiple trunks with, rather > than just adding more channels to the individual trunk. I have discussed > the matt

Re: [asterisk-users] No rtpmap codec info in 200 OK

2011-12-19 Thread William Scott
On 20 December 2011 14:22, Bruce B wrote: > Can you register with Eyebeam to VSP and have it work? Make sure you are on > the exact same network as the ATA when making this test. This should isolate > the NAT issue. > Great tip. Eyebeam dosen't send a rtpmap for known codecs unless you select th

Re: [asterisk-users] India Telecom regulations

2011-12-19 Thread Raj Mathur (राज माथुर)
On Tuesday 20 Dec 2011, khalid touati wrote: > Thank you Raj, > so with VOIP license calls can go beyond our pbx to PSTN (india), > right, if so this what i needed to know to call Indian cellphone > from US (or other countries) If your objective is to originate calls in the US (using whatever te

Re: [asterisk-users] India Telecom regulations

2011-12-19 Thread khalid touati
Thank you Raj, so with VOIP license calls can go beyond our pbx to PSTN (india), right, if so this what i needed to know to call Indian cellphone from US (or other countries) On Mon, Dec 19, 2011 at 10:03 PM, Robert-IPhone wrote: > Right check out Cordia.LT > > > Sent from my iPhone 4S > > On De

Re: [asterisk-users] No rtpmap codec info in 200 OK

2011-12-19 Thread Bruce B
Can you register with Eyebeam to VSP and have it work? Make sure you are on the exact same network as the ATA when making this test. This should isolate the NAT issue. On Mon, Dec 19, 2011 at 9:27 PM, William Scott wrote: > On 20 December 2011 12:51, Bruce B wrote: > > I could be wrong but this

Re: [asterisk-users] India Telecom regulations

2011-12-19 Thread Robert-IPhone
Right check out Cordia.LT Sent from my iPhone 4S On Dec 19, 2011, at 9:58 PM, "Raj Mathur (राज माथुर)" wrote: > On Tuesday 20 Dec 2011, Steve Edwards wrote: >> On Mon, 19 Dec 2011, Nick Khamis wrote: >>> SIP in India is illegal. >> >> What about IAX, Skype, VPN, etc? > > The only thing that

Re: [asterisk-users] India Telecom regulations

2011-12-19 Thread Raj Mathur (राज माथुर)
On Tuesday 20 Dec 2011, Steve Edwards wrote: > On Mon, 19 Dec 2011, Nick Khamis wrote: > > SIP in India is illegal. > > What about IAX, Skype, VPN, etc? The only thing that is not permitted is bridging Internet calls with the Indian PSTN. In fact, that too is allowed if you have a VoIP licence

Re: [asterisk-users] No rtpmap codec info in 200 OK

2011-12-19 Thread William Scott
On 20 December 2011 12:51, Bruce B wrote: > I could be wrong but this sounds like a NAT issue rather SIP related packet > issue. I looked at this to start with. Spent sometime comparing addresses and ports between successful and failure packets. Couldn't see any ports that weren't opened on the w

Re: [asterisk-users] No rtpmap codec info in 200 OK

2011-12-19 Thread Bruce B
I could be wrong but this sounds like a NAT issue rather SIP related packet issue. You are not receiving a response back is what I get a lot of times when my NAT is not setup properly. Call goes on for 10 or 20 second (I try the echo application and it hangs up before I get to talk) and then cuts o

Re: [asterisk-users] Limit # of inbound calls on SIP trunk

2011-12-19 Thread Steve Edwards
On Mon, 19 Dec 2011, Douglas Mortensen wrote: I have a system with FreePBX, and as far as I can tell it does not provide a means to limit the number of simultaneous inbound calls on a SIP trunk. Therefore I suspect that I’ll need to do some manual dialplan manipulation. The GROUP() and GROUP

[asterisk-users] Limit # of inbound calls on SIP trunk

2011-12-19 Thread Douglas Mortensen
Hello all, I have a system with FreePBX, and as far as I can tell it does not provide a means to limit the number of simultaneous inbound calls on a SIP trunk. Therefore I suspect that I'll need to do some manual dialplan manipulation. Essentially I will have 1 (or possibly 2) SIP trunk(s) conf

Re: [asterisk-users] No rtpmap codec info in 200 OK

2011-12-19 Thread William Scott
> It seems quite unlikely that the presence of > an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have any > problems. Thanks for the reply. I'll expand on the scenario... This particular ATA does not send 'a=rtpmap' for any codec. When talking to a Asterisk PBX everything w

[asterisk-users] Use different local IP for each SIP trunk

2011-12-19 Thread Douglas Mortensen
Hello, I have a SIP provider whom I may want to have multiple trunks with, rather than just adding more channels to the individual trunk. I have discussed the matter with them & they have told me that the only way that they identify which trunk should be used for each call is simply by the sour

Re: [asterisk-users] India Telecom regulations

2011-12-19 Thread Steve Edwards
On Mon, 19 Dec 2011, Nick Khamis wrote: SIP in India is illegal. What about IAX, Skype, VPN, etc? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline

Re: [asterisk-users] India Telecom regulations

2011-12-19 Thread Nick Khamis
SIP in India is illegal. Nick. On Mon, Dec 19, 2011 at 3:06 PM, khalid touati wrote: > Hi All, > Because I am pretty sure we have people in this DL from India, I was hoping > to get the 100% accurate information, is it legal to make calls from any > coutry to Indian mobile phones through an Aste

Re: [asterisk-users] Asterisk 1.6.2.22 Now Available

2011-12-19 Thread Jeremy Kister
On 12/19/2011 4:08 PM, Asterisk Development Team wrote: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22 or for the non-404-version: http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ChangeLog-1.6.2.22 ;p -- Jeremy Kister http://jeremy.kister.net./ --

[asterisk-users] Asterisk 1.6.2.22 Now Available

2011-12-19 Thread Asterisk Development Team
The Asterisk Development Team has announced the release of Asterisk 1.6.2.22. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sample related to AST-2011-013: * The sample

[asterisk-users] India Telecom regulations

2011-12-19 Thread khalid touati
Hi All, Because I am pretty sure we have people in this DL from India, I was hoping to get the 100% accurate information, is it legal to make calls from any coutry to Indian mobile phones through an Asterisk server based in India? -- Khalid Touati Network Administrator --

[asterisk-users] A lot of 603 Declined Error form iCall - Are they going down or is it just bad service?

2011-12-19 Thread Bruce B
Hi everyone, Since three weeks ago, we have been getting A LOT of 603 Declined calls from iCall. I called a few times and their support is either non-responsive (they never call back) or can't fix the issue. I am wondering if everyone else is experiencing the same thing or is it because we recentl

Re: [asterisk-users] Dahdi 2.5.0.2 - Strange Warning

2011-12-19 Thread Richard Mudgett
> On a recently updated system , I'm now reading lines as this one > (never noticed them before): > > [Dec 19 19:01:52] WARNING[10828]: chan_dahdi.c:11302 pri_dchannel: > Ring requested on unconfigured channel 0/0 span 4 > > My setup is: > Asterisk 1.6.1.18 > Libpri 1.4.12 > Dahadi 2.5.0.2 > > M

[asterisk-users] Which Dahdi/Libpri version are you using ?

2011-12-19 Thread Olivier
Hi, I've recently met weird behaviour on 2 different and newly upgraded libpri1.4.12/2dahdi2.5 systems (at the moment, I can't correctly describe the symptoms but that's another story). For various reasons, this lead me to wonder which Dahdi/Libpri combination/version is the most widely used on t

[asterisk-users] ChanSpy in whisper mode - low quality audio

2011-12-19 Thread ISABEL ORDAS ARNAL
Hi all, I succeed in injecting audio into one channel by mean of ChanSpy, but this audio cannot be listened correctly. I am using softphones for Android and iPhone and there are so many cuts so that they cannot understand what is said in the audio file. Is this because the total RTP bandwidth i

[asterisk-users] Dahdi 2.5.0.2 - Strange Warning

2011-12-19 Thread Olivier
Hi, On a recently updated system , I'm now reading lines as this one (never noticed them before): [Dec 19 19:01:52] WARNING[10828]: chan_dahdi.c:11302 pri_dchannel: Ring requested on unconfigured channel 0/0 span 4 My setup is: Asterisk 1.6.1.18 Libpri 1.4.12 Dahadi 2.5.0.2 My card is a Junghan

Re: [asterisk-users] Sending Fax Dialplan with Retry Attempt

2011-12-19 Thread Zoel Hairi
> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Stefan Schmidt > Sent: Monday, December 19, 2011 8:32 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Sendin

Re: [asterisk-users] Sending Fax Dialplan with Retry Attempt

2011-12-19 Thread Zoelhairi
sorry. i put the wrong the dialplan. it already RetryAttempt2 in it.  exten => s,n(RetryAttempt3),Wait(6) exten => s,n,NoOp( SENDING FAX RETRY ATTEMPT 2 : ${FAXFILE} : ${FAXOPT(error)} ) exten => s,n,SendFAX(${FAXFILE}) ;zoel : Add Retry Attempt 3 exten => s,n,GotoIf($["${FAXOPT(error)

Re: [asterisk-users] No rtpmap codec info in 200 OK

2011-12-19 Thread Kevin P. Fleming
On 12/19/2011 07:15 AM, William Scott wrote: Ok. Asterisk sends the rtpmap info for the codec. Is it possible to remove this from the 200 OK sent by Asterisk? Possible direction I should look. No, it is not, at least not without patching the Asterisk source code (which of course you are free

Re: [asterisk-users] Sending Fax Dialplan with Retry Attempt

2011-12-19 Thread Stefan Schmidt
Am 19.12.11 14:26, schrieb Zoel Hairi: > Hello All, > > > > I have a problem with Fax For Asterisk, the Successful Rate when sending Fax > are very Low especially when we send the Fax just once. Now I’m trying to > modify the dialplan so it will keep trying to send the fax for maximum 5 > ti

[asterisk-users] Sending Fax Dialplan with Retry Attempt

2011-12-19 Thread Zoel Hairi
Hello All, I have a problem with Fax For Asterisk, the Successful Rate when sending Fax are very Low especially when we send the Fax just once. Now I’m trying to modify the dialplan so it will keep trying to send the fax for maximum 5 times at once and it only retry if the Sending Status has

Re: [asterisk-users] No rtpmap codec info in 200 OK

2011-12-19 Thread William Scott
Ok. Asterisk sends the rtpmap info for the codec. Is it possible to remove this from the 200 OK sent by Asterisk? Possible direction I should look. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t

Re: [asterisk-users] AMI and Dialplan

2011-12-19 Thread --[ UxBoD ]--
Please ignore as this was a user error! -- Thanks, Phil - Original Message - > Hello all, > This may sound an odd question but if you initiate a call using AMI > does it adhere to what has been defined in the dial plan or do we > have to write the logic into the AMI call ? > -- > Th

[asterisk-users] AMI and Dialplan

2011-12-19 Thread --[ UxBoD ]--
Hello all, This may sound an odd question but if you initiate a call using AMI does it adhere to what has been defined in the dial plan or do we have to write the logic into the AMI call ? -- Thanks, Phil -- _ -- Bandwidth

Re: [asterisk-users] Play audio file for both Caller and Callee in a call

2011-12-19 Thread ISABEL ORDAS ARNAL
Hi all, I made it easier, AMI was not required, it can be solved directly in the dialplan: same => n,Dial(SIP/${TRUNK}/${ARG2}${NUM},60,M(inject^${CALLERNUMBER})) [macro-inject] same => s,1,Originate(Local/trunk@injectWarning,app,Playback,Message-Callee) same => n,Originate(Local/${ARG1}@inject