Re: [asterisk-users] Call audio leaking between calls
I agree with the cross talk analysis. My suggestion would be to focus your efforts on the analog trunks/stations, not SIP. Are you using twisted pair or shielded cables for your analog runs? If not, you might consider changing the cables or at least increasing the physical distance between them - in my experience this is the most common cause for cross talk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Tuesday, November 10, 2009 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call audio leaking between calls Doug Lytle wrote: Ishfaq Malik wrote: Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of It's called cross talk and yes, we've experienced it. But, it will only happen on an analog network (PSTN). At that point, the provider had to check the analog lines. It eventually was fixed. In a purely SIP environment, you shouldn't see this. Doug This is what I'm thinking too and it's a weird one to try to pin down, especially as I've currently got very little information. I think I'm going to use Monitor on all their calls and see if the recordings show any signs of this cross talk but even if they do it still doesn't help to resolve the issue. You'd think it would be an impossibility due to the nature of IP traffic. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ring All Queue
Is there a way in the dialplan to figure out which agent in a ring all queue answered a line? I'd like to take specific action based on the agent upon hangup. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siemens HiPath HG1500
Has anyone successfully gotten a HiPath system to route calls over to a * box? If so, I'd appreciate a quick consult. I've configured the HG card to look for the * server but it doesn't seem to actually be connecting. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simple Call Screener
I'm trying to build a simple accept/reject screening app for inbound calls that
* forwards to my cell phone. Basically I want * to announce the caller ID and
then let me press 1 to accept the call or 2 to reject the call and send the
outside party to voicemail.
I've been messing around with variation of the script below... can anyone tell
me what I'm doing wrong? It's got to be something obvious that I've overlooked.
Thanks!!!
[main]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Wait(1)
exten = s,n,Dial(SIP/[EMAIL PROTECTED],120,gM(screen))
exten = s,n,PlayBack(vm-goodbye)
exten = s,n,Hangup
[macro-screen]
exten = s,1,Wait(1)
;exten = s,n,SayDigits(${CALLERID(num)})
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=30)
exten = s,n,Background(accept-reject)
exten = 1,1,Set(MACRO_RESULT=CONTINUE)
exten = 2,1,PlayBack(vm-goodbye)
exten = 2,2,Hangup
exten = s,6,Wait(10)
exten = i,1,Goto(TT_VO,s,1)
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando, FL 32801
(407) 517-3105 - Direct Telephone
(407) 839-0120 - Main Office
(407) 841-9726 - Fax
http://www.rissman.com/
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[asterisk-users] Temporary Service - Dominican Republic DID
Hello List, I'm trying to help a family from the Dominican Republic and to do so need a temporary DID from DR. The short story is that there is a 2 year old here with a serious heart defect from a remote area of Dominican Republic near the Haitian border. He was referred to Gift of Life D.R. who eventually contacted Gift of Life Central Florida and this is where I'm involved. The child and mother are here in the US and the child had surgery last week. I need a DR DID to help facilitate phone calls home to the father in DR who couldn't make the trip here. This is a short term need and would help make a significant impact on this family. If you have a DID available, please let me know. I don't mind paying for the service but I'm having difficulty quickly in locating a provider. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay in processing dial string
We have an issue with Linksys SPA2102-NA ATA's where there is a several second delay between when you finish dialing and when it sends the commands on to *. Has anyone else seen this before? If so, is there a quick/easy solution? Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Program Closes
Periodically, maybe once or twice every few weeks, we see our instance of Asterisk 1.4.7 just close out without warning and we have to reload the module. We're running CentOS. Has anyone else seen this before? Core show version: Asterisk 1.4.7 built by root @ XX on a i686 running Linux on 2007-07-11 00:21:57 UTC Uname -a: Linux XX 2.6.9-55.0.2.EL #1 Tue Jun 26 14:08:18 EDT 2007 i686 athlon i386 GNU/Linux Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on Hold -- Error
We use Asterisk 1.4.7 on CentOS with Bandwidth.com as our provider and Polycom 330's for endpoints. When one of our end points places a call on hold we get the following in CLI. There is no music on hold provided for the caller. The SIP.CONF entry for our connection to bandwithd.com specifies disallow=all and allow=ulaw. Should there be a similar setting on the user.conf entries? An interesting note is the IP noted in the CLI message below is neither Bandwidth.com nor the end point. Thanks for any help!! CLI Message: [Nov 15 13:21:38] WARNING[11327]: channel.c:2964 set_format: Unable to find a codec translation path from ulaw to unknown [Nov 15 13:21:38] WARNING[11327]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/4.68.250.148-08d1e7e0' to format 'unknown' Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold -- Error
Interesting. Is the upgrade difficult? I've not attempt to upgrade our production environment yet. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Brown Sent: Thursday, November 15, 2007 7:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold -- Error I posted to the list earlier this week about this very issue. This reinforces my thought that it is a bug in 1.4.7. Since upgrading the box to 1.4.13 the issue resolved itself. I have not opened a issue in the tracker as I hadn't had time to try and replicate the issue. On 16/11/07 5:32 AM, Ryan M. Colbert [EMAIL PROTECTED] wrote: We use Asterisk 1.4.7 on CentOS with Bandwidth.com as our provider and Polycom 330's for endpoints. When one of our end points places a call on hold we get the following in CLI. There is no music on hold provided for the caller. The SIP.CONF entry for our connection to bandwithd.com specifies disallow=all and allow=ulaw. Should there be a similar setting on the user.conf entries? An interesting note is the IP noted in the CLI message below is neither Bandwidth.com nor the end point. Thanks for any help!! CLI Message: [Nov 15 13:21:38] WARNING[11327]: channel.c:2964 set_format: Unable to find a codec translation path from ulaw to unknown [Nov 15 13:21:38] WARNING[11327]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/4.68.250.148-08d1e7e0' to format 'unknown' Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ http://www.rissman.com/http://www.rissman.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Nick Brown Ipera Communications Pty Ltd Level 1, 9 Denison Street, Newcastle West NSW 2302 PO Box 2115, Dangar NSW 2309 Ü P: +61 2 4910 1000 Ü F: +61 2 4910 1099 Ü ABN: 31 090 964 104 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech Rec on Voicemail
I'd be interesting in pooling resources for this. We've seen the success of Vonage's Visual Voicemail and would like to emulate a similar solution. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Friday, August 24, 2007 1:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Speech Rec on Voicemail Nuance offers an SDK to do something similar, I think they say you can only expect between 45-60% accuracy using it though. Total cost is about $6K to $8K for one server license. If there are enough people interested in pooling money I'd be willing to help set up a system to process voicemails and provide the Nuance converted transcript. However, I figure the low accuracy would be the biggest turn off from using Nuance. On 8/23/07, Stephen Bosch [EMAIL PROTECTED] wrote: Ryan M. Colbert wrote: I've had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? I seem to recall that Lt. Cmdr Jordi Laforge did some stuff like this not too long ago. I get requests like this all the time -- but the technology is very far from being there. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speech Rec on Voicemail
I've had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? Thanks! Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
