[asterisk-users] Problem with outbound dialing from remote phone
I have a real head scratcher . . . We have several employees who work from home. All have Polycom 501's that register to our office Asterisk 1.6.x server and communicate using SIP g729a. About two weeks ago, one of these remote users starting experiencing a problem with a previously working phone: a. She could receive inbound calls, b. She can place outbound calls to internal extensions c. She cannot place outbound calls to external destinations. I brought up the Asterisk CLI and had her dial outbound while I watched. The calls to internal extensions are processing as they should. However, I do not see the external dial attempts ever getting to the server. This is odd because there is absolutely nothing in the programming of the phone that distinguishes one from the other. I had her key in several strings on nonsense and I saw some, but not all of them. So, I programmed another phone, and tested it thoroughly from my own remote location. Phone works fine inbound and outbound. I then shipped the phone to the user. User received new phone, plugged it in. It registers to the Asterisk server just fine. It receives inbound calls, however this one cannot dial out at all. I see no dial attempts whatsoever on CLI. If she plugs the old phone back in, she can still dial internal extensions. I know the problem is not with the phone dial pattern, as I've had her key in the number and then press the Dial key. Besides, the phone worked 24 hours earlier from a different location. The sip.conf configuration has not changed from when the phone worked properly: [1234] type=friend regext=1234 context=longdistance secret=* callerid=User Name 1234 host=dynamic qualify=yes mailbox=1234 permit=0.0.0.0/0.0.0.0 I've checked all log files, and for the failed attempts I see nothing ever getting to the server. I don't think the problem is with the phone. Any ideas, suggestions, etc., would be greatly appreciated. If I need to provide additional info please advise. Thanks. The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
I wish it was that easy. That is one of the first things we tried. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: Any ideas, suggestions, etc., would be greatly appreciated My guess that the Polycom digitmap isn't being loaded (sip.cfg). I'm sure if she were to dial the phone number and then press 'send' soft key, it'd probably dial. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
No change, thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 14, 2011 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone What happens if she keys in the number+# then presses dial? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with outbound dialing from remote phone I have a real head scratcher . . . We have several employees who work from home. All have Polycom 501's that register to our office Asterisk 1.6.x server and communicate using SIP g729a. About two weeks ago, one of these remote users starting experiencing a problem with a previously working phone: a. She could receive inbound calls, b. She can place outbound calls to internal extensions c. She cannot place outbound calls to external destinations. I brought up the Asterisk CLI and had her dial outbound while I watched. The calls to internal extensions are processing as they should. However, I do not see the external dial attempts ever getting to the server. This is odd because there is absolutely nothing in the programming of the phone that distinguishes one from the other. I had her key in several strings on nonsense and I saw some, but not all of them. So, I programmed another phone, and tested it thoroughly from my own remote location. Phone works fine inbound and outbound. I then shipped the phone to the user. User received new phone, plugged it in. It registers to the Asterisk server just fine. It receives inbound calls, however this one cannot dial out at all. I see no dial attempts whatsoever on CLI. If she plugs the old phone back in, she can still dial internal extensions. I know the problem is not with the phone dial pattern, as I've had her key in the number and then press the Dial key. Besides, the phone worked 24 hours earlier from a different location. The sip.conf configuration has not changed from when the phone worked properly: [1234] type=friend regext=1234 context=longdistance secret=* callerid=User Name 1234 host=dynamic qualify=yes mailbox=1234 permit=0.0.0.0/0.0.0.0 I've checked all log files, and for the failed attempts I see nothing ever getting to the server. I don't think the problem is with the phone. Any ideas, suggestions, etc., would be greatly appreciated. If I need to provide additional info please advise. Thanks. The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
Turned on sip set debug peer 1234. I see the qualify messages. I see when she calls me on my internal extension. I see no SIP messages at all when she calls my cell phone. I understand what Doug and Eric are saying. I need to get into the phone's web interface to see how it is programmed just to validate that the phone is still as I programmed it. What is strange is: a. Phone A can dial local extensions but not external, so I send her Phone B. b. Phone B cant dial outbound at all c. Both phones were successfully tested for both call types prior to shipping and were not in any way reconfigured subsequent to testing. d. I have not modified the digitmap is sip.cfg in years, and even so, entering the number and then pressing 'Dial' doesn't work either. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Friday, October 14, 2011 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Hey, Can you enable sip trace for that particular sip extension. This sounds weird that while other INVITES from the phone are reaching but the external extensions are filtered. If there are no invites for external calls only then more chances are that the phone is using some dial pattern(phonebook help) etc like Doug and Eric said. Sometimes in asterisk console I don't see anything in logs if the Sip extensions' context don't contain the number that is being dialled Do you've access to any phone debugging console? Sounds like problem is somewhere around She :p j/k . -- Regards, Sammy. On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins arob...@pharmacentra.commailto:arob...@pharmacentra.com wrote: The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 The information contained in this transmission may contain privileged and confidential
Re: [asterisk-users] Problem with outbound dialing from remote phone
Thanks I will do that. The user is remote, so I must first RDP into her home network and do it from her PC. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 14, 2011 3:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I use 501's here and I can pull up the settings by typing http://1.2.3.4/index.htm - where 1.2.3.4http://1.2.3.4/index.htm%20-%20where%201.2.3.4 is the IP address of the phone. If you can do that, perhaps something there will be of use to you. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Turned on sip set debug peer 1234. I see the qualify messages. I see when she calls me on my internal extension. I see no SIP messages at all when she calls my cell phone. I understand what Doug and Eric are saying. I need to get into the phone's web interface to see how it is programmed just to validate that the phone is still as I programmed it. What is strange is: a. Phone A can dial local extensions but not external, so I send her Phone B. b. Phone B cant dial outbound at all c. Both phones were successfully tested for both call types prior to shipping and were not in any way reconfigured subsequent to testing. d. I have not modified the digitmap is sip.cfg in years, and even so, entering the number and then pressing 'Dial' doesn't work either. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Friday, October 14, 2011 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Hey, Can you enable sip trace for that particular sip extension. This sounds weird that while other INVITES from the phone are reaching but the external extensions are filtered. If there are no invites for external calls only then more chances are that the phone is using some dial pattern(phonebook help) etc like Doug and Eric said. Sometimes in asterisk console I don't see anything in logs if the Sip extensions' context don't contain the number that is being dialled Do you've access to any phone debugging console? Sounds like problem is somewhere around She :p j/k . -- Regards, Sammy. On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins arob...@pharmacentra.commailto:arob...@pharmacentra.com wrote: The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun
Re: [asterisk-users] Problem with outbound dialing from remote phone
I appreciate this real out-of-the-box thinking! I can just wait for her to go to dinner and get into her PC. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, October 14, 2011 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Un-top-posting and trimming cruft... On Fri, 14 Oct 2011, Adam Robins wrote: Thanks I will do that. The user is remote, so I must first RDP into her home network and do it from her PC. Since you have access to the Asterisk server's command line, you could use wget to retrieve the index page from the phone outputting to a file and scp the file back to you for your local viewing pleasure. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Problem with outbound dialing from remote phone
I'm working that angle. I tried to use Dameware to get into her router via her home PC, but the screens weren't drawing correctly. I'll need to try LogmeIn. Also the IP address she read me directly off the phone is dubious. I cant ping it nor can I bring up the web interface. To be continued . . . -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Bosiljevac Sent: Friday, October 14, 2011 5:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Did anything else change on her home network that could correlate to the time this started flaking on you? (eg: a new router/gateway) BB -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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 detection in dialplan - Asterisk 1.6
We have an employee who works from home. We sent her a SIP phone to work as an extension off our Asterisk 1.6 system, but her DSL service is so bad she was dropping calls all the time. It's not just a tuning or QoS issue. Her service is simply unreliable. She had a POTS line installed and I have the dialplan set up so that when her extension is dialed, it calls out over our SIP provider to her 10-digit POTS number. If she is on the phone and her line is busy, I want Asterisk to place the caller into her Asterisk voicemail rather than hearing a busy signal. The way I have this working currently is by using Followme without a preceding Dial command. Seems that the Followme app handles the busy properly. The problem is that every call she receives is announced and requires her to press 1 to accept or 2 to reject. I suppose I could modify the Followme code, but I'd rather not. Any ideas are appreciated. Thanks. The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Busy detection in dialplan - Asterisk 1.6
Didn't work. It correctly times out after 20 seconds and continues to voicemail, but the caller still hears the remote busy signal during those 20 seconds. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, October 21, 2010 9:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Thursday, October 21, 2010 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 We have an employee who works from home. We sent her a SIP phone to work as an extension off our Asterisk 1.6 system, but her DSL service is so bad she was dropping calls all the time. It's not just a tuning or QoS issue. Her service is simply unreliable. She had a POTS line installed and I have the dialplan set up so that when her extension is dialed, it calls out over our SIP provider to her 10-digit POTS number. If she is on the phone and her line is busy, I want Asterisk to place the caller into her Asterisk voicemail rather than hearing a busy signal. The way I have this working currently is by using Followme without a preceding Dial command. Seems that the Followme app handles the busy properly. The problem is that every call she receives is announced and requires her to press 1 to accept or 2 to reject. I suppose I could modify the Followme code, but I'd rather not. Any ideas are appreciated. Thanks. I know how this works with DAHDI/POTS; don't know what it will do dialing over SIP Exten = 1234,1,Dial(DAHDI/1/w5551212,20,KkTt) Exten = 1234,n,voicemail(1...@default) Exten = 1234,n,hangup Exten = 1234-BUSY,1,voicemail(1...@default) Exten = 1234-CONGESTION,1,voicemail(1...@default) When I dial 1234, the other side has 20 seconds (about 4 rings) to pick up. If no pickup, voicemail is called. Lines 4 and 5 might (or might not) be redundant -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] Busy detection in dialplan - Asterisk 1.6
This did the trick! Masks the busy signal. Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, October 21, 2010 1:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 Try changing KkTt to rKkTt. This should generate a phony ring until the call is picked up or stops. -- _ -- 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 The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- 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] How do I create an IVR/Dial Group that worksproperly?
Have you tried replacing the s extension with _x.?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord (News)
Sent: Friday, July 17, 2009 11:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
On 17/07/09 14:14, Danny Nicholas wrote:
I may 100% off here, but I seem to recall reading in the last 2 days threads
that macro dialing messes with CDR entries. I would try replacing one of
your macro lines with a straight Dial command to verify this.
Thanks Danny, but that doesn't really help. I have tried moving the
contents of the offending Macro into the IVR menu itself and using a
Dial() command. But it makes no difference. The call is still on the s
extension and the CDR records the connection with the correct callerid
but with the destination as s. Which is not what I want.
If the caller dials an extension number, say 101, then it all works
fine. The problem is when trying to automatically dial from within the
plan it fails. I need to somehow change s to the end extension number
of the person who actually picks up the phone.
I am trying to understand how other people configure their * to achieve
the requirement I specified below.
I can't believe it is this hard to do. But I fail to see how I can
achieve it, because there is no extension - other than s - when the
caller enters the dialplan. I want the caller to be automatically
connected to one or other of our extensions if they do not know the
extension number to dial themselves.
I guess I am trying to find out if I have set this up totally *wrong*
and perhaps I should be using a queue or something, but that seems a bit
overkill...
Alan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 3:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
If the incoming caller dials 101 whilst in the IVR, the log is correct.
I can see *why* I am having this problem (There is no extension when you
arrive in the IVR other than s), but I cannot see *how* to fix it.
Please can I ask how do others handle this so it works properly (I've
included the basics of my DP below)?
I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
Thanks
Alan
Here is the IVR which callers are dropped into:
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten = s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten = s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten = _s-.,1,Goto(s-NOANSWER,1)
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Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
no
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tracinet
Sent: Friday, March 06, 2009 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins arob...@pharmacentra.com
wrote:
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade.
On the server that SENDS the call, I have the following in SIP.CONF:
[192.168.1.2_OB]
type=peer
fromuser=OB
host=192.168.1.2
And in EXTENSIONS.CONF
exten = 91NXXNXX,1,Dial(SIP/${ext...@192.168.1.2_ob)
On the RECEIVING Server in SIP.CONF:
[OB]
type=user
context=longdistance
I am not using a REGISTER statement on the receiving server.
My problem is that the only way I can seem to get the call delivered
into the proper SIP context on the receiving box is to use the
fromuser=OB on the sending machine. I tried using username=OB, but
then it delivers into the default context. I don't want to use
fromuser because it overrides the callerid.
Any suggestions?
Thanks,
Adam
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Did you ever get a resolution on this?
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[asterisk-users] benchg729 - no valid g729 license
I have five Asterisk servers running 1.2.14, and am planning to upgrade to 1.4 this weekend. In preparation, to use the most efficient g729 codec, I am running the new benchg729 program. It works great on two systems, but on the other three it says it cannot locate a valid g729 license. I have valid licenses on all systems, which show just fine when typing show g729 from CLI. Any ideas are appreciated. ___ -- 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] benchg729 - no valid g729 license
That did it. Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, February 18, 2009 4:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] benchg729 - no valid g729 license Adam Robins wrote: I have five Asterisk servers running 1.2.14, and am planning to upgrade to 1.4 this weekend. In preparation, to use the most efficient g729 codec, I am running the new benchg729 program. It works great on two systems, but on the other three it says it cannot locate a valid g729 license. I have valid licenses on all systems, which show just fine when typing show g729 from CLI. How recently have you re-run the 'register' tool for those licenses? It's possible the license files are in an old format that the new programs don't expect. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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
Re: [asterisk-users] Voicemail post-processing
Thanks for the suggestions. Modifying the sendmail command in voicemail.conf sounds like the most straightforward method, however, I will first try using 'record' in the dialplan instead of calling voicemail. This is so I can control the naming of the recorded file. I will simply run my externnotify script from the hangup priority to encrypt and email the file. Another drawback to using voicemail, in any form, is that when the mail recipient unencrypts the file, all voicemail recordings are names msg000x.wav. It is much better if I can name the files like vm-calleridnum-timestamp.wav Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Thursday, February 05, 2009 6:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail post-processing On Thu, Feb 05, 2009 at 05:04:11PM -0500, Adam Robins wrote: I have an application where a caller leaves a voicemail message and then I need to gpg encrypt the file before emailing it. I wrote a perl script to do this, which is executed after a message is left, using the externnotify feature in voicemail.conf. Why not abuse the sendmail command parameter of voicemail.conf and send commands through a wrapper script that handles GPG encryption? -- 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 -- 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] Voicemail post-processing
I have an application where a caller leaves a voicemail message and then I need to gpg encrypt the file before emailing it. I wrote a perl script to do this, which is executed after a message is left, using the externnotify feature in voicemail.conf. My script has no knowledge of the name of the voicemail wav file created by Asterisk (msg000x.wav). So, I retrieve a list of all files in the directory and then process all the ones I find. The problem is that if another caller is leaving a message at the same time, I am inadvertently pulling that file too, and end up emailing it as a corrupt wav file. Even if I call the script from the dialplan (at the hangup priority) instead of using externnotify, I think I would face the same issue because I cannot control the naming of the voicemail wav file. If I could select the name of the file to be written, or query the Asterisk-selected name as a system variable after the VoiceMail command is executed, I'd be ok. Any suggestions are appreciated. Thanks. ___ -- 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] Dropping incompatible voice frame
Thanks, placing: Disallow=all Allow=ulaw In the specific iaxy device context fixed it. I had always thought that allowing all possible valid codecs under the general context would work and the devices would sort it out upon handshake. Guess not. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven J. Douglas Sent: Wednesday, January 28, 2009 9:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dropping incompatible voice frame Don't use g729 in the iax.conf for the IAXY device. It doesn't support it. Regards, Steve Adam Robins wrote: I am using a Polycom SIP phone (ext 2042) to call an analog phone connected via an IAXY (ext 2120). The analog phone rings, and when I answer, I can hear the person speaking on the SIP phone, but they cannot hear me. However, if I originate the call from the analog phone to the SIP phone, it works just fine. In SIP.conf: Disallow=all Allow=g729 Allow=ulaw Canreinvite=no In IAX.conf: Disallow=all Allow=ulaw Allow=g729 Transfer=no Codecpriority=host CLI shows: [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Executing [2...@international:1] Dial(SIP/2042-b7b0cc88, IAX2/2120|12|oWwtT) in new stack [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Called 2120 [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Call accepted by 192.168.2.61 (format ulaw) [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Format for call is ulaw [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- IAX2/2120-3849 is ringing [Jan 28 16:04:33] VERBOSE[3428] logger.c: [Jan 28 16:04:33] -- IAX2/2120-3849 answered SIP/2042-b7b0cc88 [Jan 28 16:04:33] NOTICE[3428] channel.c: Dropping incompatible voice frame on IAX2/2120-3849 of format g729 since our native format has changed to 0x4 (ulaw) [Jan 28 16:04:41] VERBOSE[3428] logger.c: [Jan 28 16:04:41] -- Hungup 'IAX2/2120-3849' This is Asterisk 1.4.22, but it also happened on 1.2.4. If I call an IAX2/ulaw softphone from the SIP phone, it works fine. Could it be something in the IAXY provisioning? Any ideas are appreciated. Thanks. ___ -- 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 ___ -- 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] Dropping incompatible voice frame
I am using a Polycom SIP phone (ext 2042) to call an analog phone connected via an IAXY (ext 2120). The analog phone rings, and when I answer, I can hear the person speaking on the SIP phone, but they cannot hear me. However, if I originate the call from the analog phone to the SIP phone, it works just fine. In SIP.conf: Disallow=all Allow=g729 Allow=ulaw Canreinvite=no In IAX.conf: Disallow=all Allow=ulaw Allow=g729 Transfer=no Codecpriority=host CLI shows: [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Executing [2...@international:1] Dial(SIP/2042-b7b0cc88, IAX2/2120|12|oWwtT) in new stack [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Called 2120 [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Call accepted by 192.168.2.61 (format ulaw) [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Format for call is ulaw [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- IAX2/2120-3849 is ringing [Jan 28 16:04:33] VERBOSE[3428] logger.c: [Jan 28 16:04:33] -- IAX2/2120-3849 answered SIP/2042-b7b0cc88 [Jan 28 16:04:33] NOTICE[3428] channel.c: Dropping incompatible voice frame on IAX2/2120-3849 of format g729 since our native format has changed to 0x4 (ulaw) [Jan 28 16:04:41] VERBOSE[3428] logger.c: [Jan 28 16:04:41] -- Hungup 'IAX2/2120-3849' This is Asterisk 1.4.22, but it also happened on 1.2.4. If I call an IAX2/ulaw softphone from the SIP phone, it works fine. Could it be something in the IAXY provisioning? Any ideas are appreciated. Thanks. ___ -- 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] SIP Callerid Question
I have two Asterisk 1.2 boxes across a WAN. Calls between them are sent via SIP g729a. The issue is that the original calleridnum is overwritten by the value of the fromuser parameter in sip.conf on the originating server. Is there any way to preserve the original calleridnum value? Calleridname is not affected. I suppose I could concatenate the number into the name field and then parse it out at the other end, but . . . I know this issue has been around for a while, and is documented. I'm wondering if anything has changed or there are any new solutions: Thanks, Adam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Callerid Question
Nevermind, I just answered my own question. Used username instead of fromuser. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Friday, August 15, 2008 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP Callerid Question I have two Asterisk 1.2 boxes across a WAN. Calls between them are sent via SIP g729a. The issue is that the original calleridnum is overwritten by the value of the fromuser parameter in sip.conf on the originating server. Is there any way to preserve the original calleridnum value? Calleridname is not affected. I suppose I could concatenate the number into the name field and then parse it out at the other end, but . . . I know this issue has been around for a while, and is documented. I'm wondering if anything has changed or there are any new solutions: Thanks, Adam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk in Xen domu with tdm400 hardware
We are running Asterisk on native CentOS. We then install VMWare on CentOS with Windows 2003 in the VMWare partition for AD services. We have 50+ users in a call center environment with no issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Creasy Sent: Sunday, May 27, 2007 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Why would you want to do this? If you wanted to run multiple systems together on an Asterisk server I would run the Asterisk server on Dom0 and the other stuff on DomU systems. -Jonathan James Harper wrote: I did it back in the xen 2.x days with a BRI adapter (Traverse NetJet). It worked fine for the testing I was doing. I'm not sure of the status or performance of the PCI mapping through to DomU these days, but that should be the only extra step required. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roberto Pereyra Sent: Saturday, 26 May 2007 23:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Hi all !!! I would like to install asterisk in Xen domU using TDM400 hardware. Somebody know a howto or tutorial about that ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk in Xen domu with tdm400 hardware
Thanks, but we do not use any zap hardware in these systems. It is straight SIP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Tuesday, May 29, 2007 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Hi, Be careful with believing too much that your zaptel hardware will work together with xen, you could have problems like the ones described in the thread linked below: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg180825.html Good luck, François. Adam Robins wrote: We are running Asterisk on native CentOS. We then install VMWare on CentOS with Windows 2003 in the VMWare partition for AD services. We have 50+ users in a call center environment with no issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Creasy Sent: Sunday, May 27, 2007 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Why would you want to do this? If you wanted to run multiple systems together on an Asterisk server I would run the Asterisk server on Dom0 and the other stuff on DomU systems. -Jonathan James Harper wrote: I did it back in the xen 2.x days with a BRI adapter (Traverse NetJet). It worked fine for the testing I was doing. I'm not sure of the status or performance of the PCI mapping through to DomU these days, but that should be the only extra step required. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roberto Pereyra Sent: Saturday, 26 May 2007 23:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Hi all !!! I would like to install asterisk in Xen domU using TDM400 hardware. Somebody know a howto or tutorial about that ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk in Xen domu with tdm400 hardware
This is why we installed Asterisk on CentOS directly and then put Windows under a VMWare partition, rather than put bot CentOS and Windows under VMWare -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cosmin Prund Sent: Tuesday, May 29, 2007 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Keep in mynd, SIP requires a stable timing source. Don't know how Xen handles timing, but with vmware you can get all sorts of issues with timing: the clock goes faster or slower then normal on multi core systems and on systems with power stepping. In my case i'm getting those timing issues on two dual core amd machines and i'm not getting timing issues on three dual-core intel machines. -- Cosmin Prund -Original Message- From: Adam Robins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 29.05.07 18:09 Subject: RE: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Thanks, but we do not use any zap hardware in these systems. It is straight SIP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Tuesday, May 29, 2007 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Hi, Be careful with believing too much that your zaptel hardware will work together with xen, you could have problems like the ones described in the thread linked below: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg180825.html Good luck, François. Adam Robins wrote: We are running Asterisk on native CentOS. We then install VMWare on CentOS with Windows 2003 in the VMWare partition for AD services. We have 50+ users in a call center environment with no issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Creasy Sent: Sunday, May 27, 2007 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Why would you want to do this? If you wanted to run multiple systems together on an Asterisk server I would run the Asterisk server on Dom0 and the other stuff on DomU systems. -Jonathan James Harper wrote: I did it back in the xen 2.x days with a BRI adapter (Traverse NetJet). It worked fine for the testing I was doing. I'm not sure of the status or performance of the PCI mapping through to DomU these days, but that should be the only extra step required. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roberto Pereyra Sent: Saturday, 26 May 2007 23:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Hi all !!! I would like to install asterisk in Xen domU using TDM400 hardware. Somebody know a howto or tutorial about that ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Asterisk-GUI problem
I just installed the Digium asterisk-gui from svn on to an asterisk 1.4 beta3 configuration. I can get to the main page, cfgbasic.html, and then log in OK, however after I log in and then each time I click on a new menu item I receive Stack overflow at line: 0. None of the data Fields on the screens populate from the config files. I am running IE7 on Win XP SP2. Any assistance is appreciated. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on virtual machine
We have a centralized infrastructure where we deploy Asterisk servers in remote call centers for authentication and transcoding. SIP g729a calls are then sent over an MPLS VPN to a central Asterisk farm, from which calls aresent/received via PRI. To avoid placing two servers in each call center, one for Asterisk and another for Windows AD services, we have been playing with VMWare. Can anyone provide their experiences in using Asterisk in a VMWare configuration? Good/bad/ugly? Thanks, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail volume adjustment
This works great, however, when I look at the full log, it says that the sendmail is executing prior to vm-audio. Any way to change this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cullin J. Wible Sent: Tuesday, June 27, 2006 8:41 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voicemail volume adjustment In voicemail.conf: externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio The attached script should increase as much as possible without clipping. Cheers, Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem Using Asterisk Call Files with Zap PRI
I have an application where I need to send outbound prerecorded
messages. The Asterisk call file process works fine if I am sending
the call via SIP or IAX, but not via ZAP over a PRI channel. The
destination device (my cell phone) never rings. The only unusual thing
I see is on the fifth line of the full log, below, channel.c: Don't
know what to do with control frame 15. It never gets to the
auto_outbound context.
Asterisk version is 1.2.4.
I appreciate any assistance you may provide.
Call File config:
Channel: ZAP/G1/1770xxx
Callerid: CompanyName 800-111-
MaxRetries: 1
Context: auto_outbound
Extension: s
Priority: 1
SetVar: file=recordingtoplay
Extensions.conf:
[auto_outbound]
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Background(${file})
exten = s,n,Hangup
Full log output:
Apr 18 16:25:10 VERBOSE[12827] logger.c: -- Attempting call on
ZAP/G1/1770xxx for [EMAIL PROTECTED]:1 (Retry 1)
Apr 18 16:25:10 DEBUG[1508] channel.c: Avoiding initial deadlock for
'Zap/95-1'
Apr 18 16:25:10 VERBOSE[12827] logger.c: -- Requested transfer
capability: 0x00 - SPEECH
Apr 18 16:25:10 DEBUG[1647] chan_zap.c: Queuing frame from
PRI_EVENT_PROCEEDING on channel 0/23 span 4 Apr 18 16:25:10
NOTICE[12827] channel.c: Don't know what to do with control frame 15
Apr 18 16:25:10 VERBOSE[1647] logger.c: -- PROGRESS with cause code
0 received
Apr 18 16:25:10 DEBUG[1647] chan_zap.c: Queuing frame from
PRI_EVENT_PROGRESS on channel 0/23 span 4 Apr 18 16:25:57 DEBUG[12827]
cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Apr 18 16:25:57 DEBUG[12827] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode,userfield) VALUES
('2006-04-18 16:25:57','\CompanyName\
800111','800111','s','auto_outbound',
'Zap/95-1','','Dial','ZAP/G1/1770xxx',0,0,'FAILED',3,'','')
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is 'CompanyName
800111'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '800111'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is 's'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is 'auto_outbound'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is 'Zap/95-1'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '(null)'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is 'Dial'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is
'ZAP/G1/1770xxx'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '2006-04-18
16:25:57'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '(null)'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '2006-04-18
16:25:57'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '0'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '0'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is 'FAILED'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is 'DOCUMENTATION'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '(null)'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '1145391910.1905'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '(null)'
Apr 18 16:25:57 DEBUG[12827] chan_zap.c: Set option AUDIO MODE, value:
ON(1) on Zap/95-1 Apr 18 16:25:57 DEBUG[12827] chan_zap.c: Hangup:
channel: 95 index = 0, normal = 117, callwait = -1, thirdcall = -1 Apr
18 16:25:57 DEBUG[12827] chan_zap.c: Not yet hungup... Calling hangup
once with icause, and clearing call Apr 18 16:25:57 DEBUG[12827]
chan_zap.c: disabled echo cancellation on channel 95 Apr 18 16:25:57
DEBUG[12827] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/95-1
Apr 18 16:25:57 DEBUG[12827] chan_zap.c: Updated conferencing on 95,
with 0 conference users Apr 18 16:25:57 DEBUG[12827] chan_zap.c: Set
option AUDIO MODE, value: OFF(0) on Zap/95-1 Apr 18 16:25:57
DEBUG[12827] chan_zap.c: disabled echo cancellation on channel 95
Apr 18 16:25:57 VERBOSE[12827] logger.c: -- Hungup 'Zap/95-1'
Apr 18 16:25:57 DEBUG[12827] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Apr 18 16:25:57 DEBUG[12827] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode,userfield) VALUES
('2006-04-18 16:25:57','','','s','default',
'**Unknown**','','','',0,0,'FAILED',3,'','')
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '(null)'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '(null)'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is 's'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is 'default'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '**Unknown**'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '(null)'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '(null)'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '(null)'
Apr 18 16:25:57 DEBUG[12827] pbx.c: Function result is '2006-04-18
16:25:57'
Apr 18
[Asterisk-Users] DUNDi with SIP
Anyone out there have a functional DUNDi configuration using SIP for the inter-Asterisk transport? I've gotten it to work with IAX2, but if I change it to SIP it does not pass the call over even though it knows where to send it. Thanks. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inter-Asterisk SIP and CalleriID
When doing an inter-Asterisk call transfer using SIP, I am using the fromuser parameter to route the call into the proper context on the receiving server. This causes the original callerid to be lost. Does anyone have any ideas how to preserve the original callerid in this scenario? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inter-Asterisk Using SIP
I am switching from IAX2 to SIP for my inter-Asterisk transport due to assorted quality issues following the 1.2.4 upgrade. On the server that SENDS the call, I have the following in SIP.CONF: [192.168.1.2_OB] type=peer fromuser=OB host=192.168.1.2 And in EXTENSIONS.CONF exten = 91NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) On the RECEIVING Server in SIP.CONF: [OB] type=user context=longdistance I am not using a REGISTER statement on the receiving server. My problem is that the only way I can seem to get the call delivered into the proper SIP context on the receiving box is to use the fromuser=OB on the sending machine. I tried using username=OB, but then it delivers into the default context. I don't want to use fromuser because it overrides the callerid. Any suggestions? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with chan_iax.c implimentation causesbad audio?
We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried G729/ilbc/ulaw. It was a total mess. We switched to SIP and instantly all problems disappeared. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.c implimentation causesbad audio? On Tuesday 21 March 2006 07:19, Matt wrote: I was going to avoid naming names :P But anyway.. yes it's asterlink. Guys seem nice enough.. and by golly.. when I switched to SIP the termination is crystal clear... so far I'm happy with the service from Asterlink... just wish I could use IAX2 oh well.. it really matters not to me HOW I get the audio stream.. just that it works and is stable. I don't know why you'd avoid naming names. Asterlink does have good service, and as I said they are a smart bunch of guys. I get troubles with my SIP registrations to them on occasion but that's it. I have absolutely no trouble recommending them to anyone. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with chan_iax.c implimentation causesbadaudio?
We upgraded all five servers to 1.2.4. We tried trunking/notrunking. End users use an IAX2 softphone on their desktop PCs. Agents are VLANed and all IAX2 traffic is QoS'd on all LAN and WAN legs. Calls flow from the agents to the local Asterisk server as IAX2/ulaw. Then they went over the IP-VPN as IAX2/g729 (we tried ilbc and straight ulaw as well). Calls get to the PSTN from the central site via PRI on TE410P cards. Point is that it worked fine for 6-9 months before the Asterisk 1.2.4 upgrade. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 10:21 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.c implimentation causesbadaudio? On Tuesday 21 March 2006 09:47, Adam Robins wrote: We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried G729/ilbc/ulaw. It was a total mess. Did you upgrade all three boxes? Did you try disabling trunking? What was your last mile solution? (i.e. what did the end-users speak into, and how did their calls get to the PSTN?) If it was to a far-end Asterisk box, what version where they running? Were you communicating using IAX2 to them too? Did they upgrade to 1.2.4 as well? I am running SVN trunk with IAX2 and SIP and have *zero* issues. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio?
All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4 tremendously IMPROVED their call quality with IAX2. Headsets are Plantronics H251N tops with DA60 USB adapters. All Desktops are at least 2.0 GHz P4 with 512MB RAM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:08 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.c implimentationcausesbadaudio? On Tuesday 21 March 2006 10:55, Adam Robins wrote: End users use an IAX2 softphone on their desktop PCs. Agents are VLANed If there were significant changes to chan_iax2 and these were not upgraded to match, this could explain the trouble. Point is that it worked fine for 6-9 months before the Asterisk 1.2.4 upgrade. Oh, I understand the point. I'm not defending a protocol change causing such breakage, I am just trying to identify why the breakage occurred when Asterisk was upgraded. Out of curiosity, which softphones do you use? What kind of interface to the user, just a cheap headset plugged into the speaker/mic on a soundcard (which soundcard? I've had trouble with some) or something fancier such as a Plantronics USB headset or bluetooth one? Regards, Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On Tuesday 21 March 2006 11:19, Adam Robins wrote: All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. Right, I wouldn't suspect otherwise. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4 tremendously IMPROVED their call quality with IAX2. I wonder what the hell is going on then, that is definitely something strange. Headsets are Plantronics H251N tops with DA60 USB adapters. All Desktops are at least 2.0 GHz P4 with 512MB RAM Thanks for the information. I feel bad for not having a good solid answer for why it's occurring. As the saying goes: I don't have an answer, but I admire the problem... -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio?
Thanks for the offer. We deleted all of our Ethereal traces once we switched to SIP. On a bad call call there were tens of thousands of checksum errors and packets out of sequence. This occurred both with and without IAX2 trunking and trunktimestamps. Complaints of poor quality were from both the agent and customer sides. Mostly cutting in and out - typical of dropped packets. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Tuesday, March 21, 2006 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On 21 Mar 2006, at 16:19, Adam Robins wrote: All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4 tremendously IMPROVED their call quality with IAX2. Headsets are Plantronics H251N tops with DA60 USB adapters. All Desktops are at least 2.0 GHz P4 with 512MB RAM I don't suppose you have an ethereal packet capture from a bad call ??? Or a description of the 'badness'? I'm doing stuff in IAX2 at the moment and might be able to spot a problem. Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: [asterisk-biz] Professional Recordings
Try Allison at theivrvoice.com. She is the voice of Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, March 08, 2006 11:06 PM To: Commercial and Business-Oriented Asterisk Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: [asterisk-biz] Professional Recordings http://www.mikesullivan.com/ http://thevoice.digium.com/ On Wed, 8 Mar 2006, Waldo Rubinstein wrote: Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival tts
Can someone tell me what I'm doing wrong here? I'm trying this from the
command prompt.
# echo Hello World | /usr/bin/text2wave -scale 1.5 -F 000 -o
/tmp/1141915933.wav
rateconv: failed to convert from 16000 to 0
doing v
#
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antoine
Megalla
Sent: Thursday, March 09, 2006 8:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Festival tts
I tried doing the same things as you to make Festival work with
Asterisk, but I had a small problem with Festival only prducing the
sound if the text was tess than 14 characters
So I used the other approach and used the text2wave utility instead (I
saw on some postings that people recommended it) and it wrols like a
charm now.
Here is the complete macro I used for TTS:
[macro-sandtts]
exten = s,1,Set(FNAME=${EPOCH})
exten = s,2,System(echo ${ARG1} |
/usr/bin/text2wave -scale 1.5 -F
000 -o /tmp/${FNAME}.wav)
exten = s,3,Playback(/tmp/${FNAME})
exten = s,4,System(rm /tmp/${FNAME}.wav)
First we creat ann (almost) unique file name Next we call the text2wave
utility with correct switches and passing the text we need to pronounce
as input to the utility.
then we playback the generated wave file.
Finally we remove the generated wave file.
Just call the macro with the text you want to say and it will work for
you.
Message: 28
Date: Thu, 9 Mar 2006 11:43:56 -
From: Steven [EMAIL PROTECTED]
Subject: [Asterisk-Users] Festival tts
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hi I have installed Festival on the same box as
asterisk and followed the
instructions to integrate it with asterisk.
Festival seems to work fine on its own performing
text to speech from the
command line or via a file.
Asterisk answers the call but there is no speech. I
can see no errors in
the
Festival log file
The asterisk console shows
--Executing Answer(SIP/81801-c091, ) in a new
stack
--Executing Festival(SIP/81801-c091, mary had a
little lamb) in a new
stack
==Parsing '/etc/asterisk/festival.conf':Found
there is nothing else after this
If I start festival as festival --server I can see
the output
Server 11:39:14 : Festival server started on port
1314
Client(1) 11:39:21 : accepted from localhost
Client(1) 11:39:21 : disconnected
Initially I added the code to festival.scm for * but
later patched the
Festival code and re-complied it.
For every test I have restarted * after Festival
Any help appreciated
Thanks
Steven
Steven Jack
Videoconferencing Manager
University of Glasgow
Computing Service
Glasgow G12 8QQ
UK
Tel +44(0)1413303828 Fax +44(0)1413303820
Email: [EMAIL PROTECTED]
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RE: [Asterisk-Users] Festival tts
No, I did not install Festival, but I saw that the text2wave module is in the usr/bin directory. I'm running RH Ent 2.4 kernel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: Thursday, March 09, 2006 10:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Festival tts Steven [EMAIL PROTECTED] wrote: Hi I have installed Festival on the same box as asterisk and followed the instructions to integrate it with asterisk. Festival seems to work fine on its own performing text to speech from the command line or via a file. Asterisk answers the call but there is no speech. I can see no errors in the Festival log file I asked the same question to this list a while back but got no replies. What OS are you using? How did you install Festival? What version of *? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival tts
I figured it out. It should read: # echo Hello World | /usr/bin/text2wave -scale 1.5 -F 8000 -o /tmp/1141915933.wav The 8 was missing in front of the 000'. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Thursday, March 09, 2006 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Festival tts No, I did not install Festival, but I saw that the text2wave module is in the usr/bin directory. I'm running RH Ent 2.4 kernel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: Thursday, March 09, 2006 10:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Festival tts Steven [EMAIL PROTECTED] wrote: Hi I have installed Festival on the same box as asterisk and followed the instructions to integrate it with asterisk. Festival seems to work fine on its own performing text to speech from the command line or via a file. Asterisk answers the call but there is no speech. I can see no errors in the Festival log file I asked the same question to this list a while back but got no replies. What OS are you using? How did you install Festival? What version of *? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I was using IAX2 with ILBC and no trunking. I also set the resyncthreshold=-1 to turn it off. Still had major jitter problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, February 23, 2006 6:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning After 2 weeks of messing around with every conceivable IAX2 and jitterbuffer configuration, I switched to SIP yesterday. Complaints went from 10-20 per day to ZERO. Literally overnight. I wonder if this is an ILBC frame size issue of some sort? Seems odd. I've got to add my name to the list here. We're just using GSM over our IAX links, and our jitterbuffer values look like this: maxjitterbuffer=1000 resyncthreshold=1000 maxjitterinterps=10 For the most part the new jitterbuffer actually yields much better quality than the old jitterbuffer, but when the resyncs happen, it's like the call has a lot of trouble getting get back on track. It flounders for quite a while, with badly broken audio, sometimes up to 20 seconds before coming back. I've tried hanging up as soon as event starts happening and then immediately calling the same number, and the channel comes back with crystal clarity. So it seems to me like there is something askew with the resync. If memory serves correctly, I believe I remember Mark applying a fix to the iax jitterbuffer and that fix had something to do with a counter rollover or something like that. That fix happened in the last week or so. I'm not sure if that would have been included in v1.2.4 or not, but might be worth a little research. I also opened a bug a month or two ago involving ilbc and iax, and someone else confirmed it was a bug. Don't have the bug number handy, but the problem related to a combination of iax trunking, jitterbuffer and ilbc. Disabling one of those consistently bypassed the problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk compile error
I'm trying to compile Asterisk 1.2.4 on a Redhat Enterprise system, kernel 2.4.21-27.0.2.ELsmp I'm getting the following errors and then the compile stops. /usr/kerberos/lib/libgssapi_krb5.so.2: undefined reference to `add_error_table'/usr/kerberos/lib/libgssapi_krb5.so.2: undefined reference to `remove_error_table'collect2: ld returned 1 exit status Can anyone point me in the right direction? I can't seem to find anything online. Thanks Adam S. RobinsExecutive Vice President CIO PHARMACENTRA,LLC 5901B Peachtree Dunwoody Road, Suite 380Atlanta, GA 30328 Office: 770-395-0088 x2034Fax: 770-395-0989Mobile: 770-855-1360Email:[EMAIL PROTECTED]Web:http://www.pharmacentra.com The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
Thanks, We already have a cron reboot of all of our Asterisk servers every night. We've been doing this for over a year due to memory leak issues. After 2 weeks of messing around with every conceivable IAX2 and jitterbuffer configuration, I switched to SIP yesterday. Complaints went from 10-20 per day to ZERO. Literally overnight. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simone Cittadini Sent: Thursday, February 23, 2006 4:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Adam Robins ha scritto: Thanks, but we already have the TOS bits set to 0xB8, which matches the QoS settings in our switches and routers. This is definitely something that changed in the 1.07 to 1.24 upgrade. We have a pair of identical 1.07 servers connected via the same network pipe that do not exhibit these issues. I might try recompiling with the old jitterbuffer to see if it makes a difference. -- -- I've not 1.24 in producton yet, still 1.21, anyway I've noticed that restarting asterisk every night dramatically reduces complaints about choppy calls (I think is something about a memory leak and not jitterbuffer, anyway is something easy to do so it's worth trying) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
It happened with g729a as well -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Thursday, February 23, 2006 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning On Feb 23, 2006, at 4:58 AM, Adam Robins wrote: Thanks, We already have a cron reboot of all of our Asterisk servers every night. We've been doing this for over a year due to memory leak issues. ??? What do you think this is windows 95??? I had a problem like that I would be looking at getting rid of asterisk. I don't ;~) I wonder what your leak is ? After 2 weeks of messing around with every conceivable IAX2 and jitterbuffer configuration, I switched to SIP yesterday. Complaints went from 10-20 per day to ZERO. Literally overnight. I wonder if this is an ILBC frame size issue of some sort? Seems odd. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I am not running trunked IAX. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Willis Sent: Monday, February 20, 2006 8:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Adam Robins wrote: This is definitely something that changed in the 1.07 to 1.24 upgrade. We have a pair of identical 1.07 servers connected via the same network pipe that do not exhibit these issues. I might try recompiling with the old jitterbuffer to see if it makes a difference. If you are running trunked IAX, try turning off the jitterbuffer entirely. Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
Title: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning This is not going over the Internet. It is going over an MPLS IP-VPN. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. LiberatoreSent: Monday, February 20, 2006 7:55 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning so you think this problem is asterisk and not a internet problem? My customers also complain alot about IAX2 connection to teliax which seemed to work better in older * versions. I have tried everything with no success, i switched to sip and its alot better but not perfect... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam RobinsSent: Monday, February 20, 2006 6:51 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Thanks, but we already have the TOS bits set to 0xB8, which matches the QoS settings in our switches and routers. This is definitely something that changed in the 1.07 to 1.24 upgrade. We have a pair of identical 1.07 servers connected via the same network pipe that do not exhibit these issues. I might try recompiling with the old jitterbuffer to see if it makes a difference. From: [EMAIL PROTECTED] on behalf of Jesus E ZepedaSent: Mon 2/20/2006 5:02 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning In my case I don't have a T1 or even a fractional T1, but cable and havenoticed that choppy calls can be reduced by adding tos settings. Like:Tos=lowdelay|throughput|reliabilityRegards,Jesus-Original Message-From: Adam Robins [mailto:[EMAIL PROTECTED]]Sent: Monday, February 20, 2006 14:43To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New JitterbufferTuningI have now set the "resyncthreshold" to -1, to turn it off. I have alsoset the "maxjitterbuffer" to 2000.I still received 10 complaints of choppy calls today on Asterisk 1.2.4versus only 1 complaint on Asterisk 1.07.-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of yusufSent: Monday, February 20, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New JitterbufferTuningAdam Robins wrote:Hi Adam After many days of playing with the new jitterbuffer and trunkingoptions for IAX2, I have finally received almost acceptable quality. Iam receiving 5-8 complaints a day of calls "breaking up" from both thecustomer and agent sides. What I have discovered is that in most ofthese cases, the new jitterbuffer performed a resync during the call.Currently, I have the resyncthreshold, and all other jb parameters attheir default levels The traffic is running over a fairly high latencyWAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping timesrun about 85ms.I am interested to know why you are using ilbc, n why not g729 ot g723or speex. What is the size of the WAN connection. How many calls areyou running over this link. I just need to see how others are fairingwith IAX2 over WAN links, as I am the final stages of testing on my sidethanks,yusuf___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersThe contents of this email message and any attachments are confidentialand are intended solely for addressee. The information may also belegally privileged. This transmission is sent in trust, for the solepurpose of delivery to the intended recipient. If you have received thistransmission in error, any use, reproduction or dissemination of thistransmission is strictly prohibited. If you are not the intendedrecipient, please immediately notify the sender by reply email anddelete this message and its attachments, if any.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use,
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
Thank you for validating that I am not going mad! I made some additional tweaks for today. We'll see how it goes. If not well, then I'll try SIP for tomorrow. Thanks, Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Fern Sent: Tuesday, February 21, 2006 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning I had exactly the same experience running IAX2, but also experienced half-duplex calls on top of that (though I think that's a different but with IAX handoff), and in the end dropped it completely for SIP. We run g729 over dedicated fibre, and the resyncs were occurring all over the place with quite ludicrous values logged for delay. I tried tweaking the jitterbuf, turning it off completely, and reverting to the old jitterbuffer implementation. none of which made any difference. I also tried with and without trunking enabled. SIP is running much more acceptably now. Adam Robins wrote: After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls breaking up from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms. Below are the resync messages for this past Friday. Knowing that I have a slow connection, should I set the resync at a much higher level? I appreciate any assistance you may provide. Thanks, Adam Feb 17 09:07:41 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -34, this delay 1651, threshold 1488, new offset -1651 Feb 17 09:07:42 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this delay -1684, threshold 1000, new offset 33 Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 176, this delay 1835, threshold 1126, new offset -1835 Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 32, this delay 1673, threshold 1062, new offset -1673 Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1663, threshold 1300, new offset -172 Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1635, threshold 1300, new offset -38 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -22, this delay 2335, threshold 1054, new offset -2373 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 11, this delay 2363, threshold 1082, new offset -2535 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -71, this delay 2249, threshold 1054, new offset -2249 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -180, this delay -2359, threshold 1360, new offset -14 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -2354, threshold 1300, new offset -181 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this delay -2297, threshold 1240, new offset 48 Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 109, this delay 1556, threshold 1136, new offset -1556 Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -30, this delay -1439, threshold 1000, new offset -117 Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1608, threshold 1048, new offset -1725 Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -29, this delay -1616, threshold 1058, new offset -109 Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 21, this delay 1751, threshold 1620, new offset -1751 Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1724, threshold 1686, new offset -1724 Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -60, this delay -1716, threshold 1000, new offset -8 Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -119, this delay -1757, threshold 1000, new offset 6 Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 75, this delay 1421, threshold 1326, new offset -1421 Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 274, this delay 1595, threshold 1282, new offset -1595 Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1311, this delay 820, threshold 1824, new offset -2415 Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1349, this delay 761, threshold 1752, new offset -2182 Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -299
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I was using G729 with Asterisk 1.07. With the new ability to do packet loss correction with ILBC, I felt I'd give it a try. The new PLC does not work with G729. I don't use Speex because my softphone does not support it. This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569 (IAX2). I've never really stressed the bandwidth. Typically, only 10-20 concurrent calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: Monday, February 20, 2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Adam Robins wrote: Hi Adam After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls breaking up from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms. I am interested to know why you are using ilbc, n why not g729 ot g723 or speex. What is the size of the WAN connection. How many calls are you running over this link. I just need to see how others are fairing with IAX2 over WAN links, as I am the final stages of testing on my side thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I have now set the resyncthreshold to -1, to turn it off. I have also set the maxjitterbuffer to 2000. I still received 10 complaints of choppy calls today on Asterisk 1.2.4 versus only 1 complaint on Asterisk 1.07. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: Monday, February 20, 2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Adam Robins wrote: Hi Adam After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls breaking up from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms. I am interested to know why you are using ilbc, n why not g729 ot g723 or speex. What is the size of the WAN connection. How many calls are you running over this link. I just need to see how others are fairing with IAX2 over WAN links, as I am the final stages of testing on my side thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
Title: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Thanks, but we already have the TOS bits set to 0xB8, which matches the QoS settings in our switches and routers. This is definitely something that changed in the 1.07 to 1.24 upgrade. We have a pair of identical 1.07 servers connected via the same network pipe that do not exhibit these issues. I might try recompiling with the old jitterbuffer to see if it makes a difference. From: [EMAIL PROTECTED] on behalf of Jesus E ZepedaSent: Mon 2/20/2006 5:02 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning In my case I don't have a T1 or even a fractional T1, but cable and havenoticed that choppy calls can be reduced by adding tos settings. Like:Tos=lowdelay|throughput|reliabilityRegards,Jesus-Original Message-From: Adam Robins [mailto:[EMAIL PROTECTED]]Sent: Monday, February 20, 2006 14:43To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New JitterbufferTuningI have now set the "resyncthreshold" to -1, to turn it off. I have alsoset the "maxjitterbuffer" to 2000.I still received 10 complaints of choppy calls today on Asterisk 1.2.4versus only 1 complaint on Asterisk 1.07.-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of yusufSent: Monday, February 20, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New JitterbufferTuningAdam Robins wrote:Hi Adam After many days of playing with the new jitterbuffer and trunkingoptions for IAX2, I have finally received almost acceptable quality. Iam receiving 5-8 complaints a day of calls "breaking up" from both thecustomer and agent sides. What I have discovered is that in most ofthese cases, the new jitterbuffer performed a resync during the call.Currently, I have the resyncthreshold, and all other jb parameters attheir default levels The traffic is running over a fairly high latencyWAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping timesrun about 85ms.I am interested to know why you are using ilbc, n why not g729 ot g723or speex. What is the size of the WAN connection. How many calls areyou running over this link. I just need to see how others are fairingwith IAX2 over WAN links, as I am the final stages of testing on my sidethanks,yusuf___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersThe contents of this email message and any attachments are confidentialand are intended solely for addressee. The information may also belegally privileged. This transmission is sent in trust, for the solepurpose of delivery to the intended recipient. If you have received thistransmission in error, any use, reproduction or dissemination of thistransmission is strictly prohibited. If you are not the intendedrecipient, please immediately notify the sender by reply email anddelete this message and its attachments, if any.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls breaking up from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms. Below are the resync messages for this past Friday. Knowing that I have a slow connection, should I set the resync at a much higher level? I appreciate any assistance you may provide. Thanks, Adam Feb 17 09:07:41 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -34, this delay 1651, threshold 1488, new offset -1651 Feb 17 09:07:42 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this delay -1684, threshold 1000, new offset 33 Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 176, this delay 1835, threshold 1126, new offset -1835 Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 32, this delay 1673, threshold 1062, new offset -1673 Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1663, threshold 1300, new offset -172 Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1635, threshold 1300, new offset -38 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -22, this delay 2335, threshold 1054, new offset -2373 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 11, this delay 2363, threshold 1082, new offset -2535 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -71, this delay 2249, threshold 1054, new offset -2249 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -180, this delay -2359, threshold 1360, new offset -14 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -2354, threshold 1300, new offset -181 Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this delay -2297, threshold 1240, new offset 48 Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 109, this delay 1556, threshold 1136, new offset -1556 Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -30, this delay -1439, threshold 1000, new offset -117 Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1608, threshold 1048, new offset -1725 Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -29, this delay -1616, threshold 1058, new offset -109 Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 21, this delay 1751, threshold 1620, new offset -1751 Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1724, threshold 1686, new offset -1724 Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -60, this delay -1716, threshold 1000, new offset -8 Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -119, this delay -1757, threshold 1000, new offset 6 Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 75, this delay 1421, threshold 1326, new offset -1421 Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 274, this delay 1595, threshold 1282, new offset -1595 Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1311, this delay 820, threshold 1824, new offset -2415 Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1349, this delay 761, threshold 1752, new offset -2182 Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -299, this delay -2127, threshold 1598, new offset -288 Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -270, this delay -2106, threshold 1540, new offset -76 Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 98, this delay 1878, threshold 1206, new offset -1878 Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 44, this delay 1799, threshold 1150, new offset -1799 Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 28, this delay 1781, threshold 1146, new offset -1781 Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1753, threshold 1000, new offset -46 Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1765, threshold 1000, new offset -16 Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -149, this delay -1747, threshold 1298, new offset -131 Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -44, this delay 1136, threshold 1064, new
[Asterisk-Users] Asterisk 1.2.4 Quality Issues
We have (had) two identical Asterisk servers for our outbound call center. Both were running Linux 2.4 kernel, Asterisk 1.0.7, Libpri 1.0.7 and Zaptel 1.2.1. Each server has a TE410P card with two PRIs. Last week, we upgraded one of them to Asterisk 1.2.4, Zaptel 1.2.3, Libpri 1.2.2. The agents on the new system suddenly started complaining that calls were cutting in/out, and that customers were having problems hearing them. We then downgraded zaptel back to 1.2.1, but no improvements. If I move the agents over to the old 1.0.7 server, they have no issues. Has anyone had similar issues? Would downgrading Libpri help anything? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Repeating Zap Message
What would cause the message: == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up To keep appearing on CLI about once every second? If I do a zap show status: Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2OK 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of *
I have done this successfully with Asterisk 1.07 and Zaptel 1.09 and 1.2.1 for the same reasons as you. However, if you ever need to go recompile Asterisk, then you will first need to recompile the old Zaptel, compile Asterisk and the new Zaptel again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Earle (CBL) Sent: Monday, January 23, 2006 4:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of * Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever) with an older version of Asterisk? I'm running 1.09, but I was wondering if I could get at the newer echo cancellers like KB1 and MG2 without upgrading to Asterisk 1.2? I'm going out on a limb here to try and fix a serious echo problem on a TDM + BT PSTN line in the UK Thanks for your suggestions everyone -- Chris Earle System Solutions Specialist, -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SAN Devices
Anyone out there using small-midsized (2-4 TB) SAN solution among multiple Asterisk systems? I don't have the budget for an EMC-caliber solution, and can't seem to find much else out there. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Dial Failover
What are you using to terminate the PSTN calls and do the SIP transcoding? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan All, I have an Asterisk system that sends PSTN calls to an OpenSER system to be routed. I have a command like this in my extensions.conf: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) There's actually two OpenSER systems for redundancy. I'm trying to find a way to have Asterisk attempt to route the call to one OpenSER system, and if it's down, fallback to another. Any first thoughts on how to achieve this? I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups are broken. If I issue a series of Dial commands, such as this: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) ... what seems to happen is that when proxy1 is down, Asterisk waits the full 20 seconds before returning control. Also, This 20s includes the time is takes for the other end to answer, so if I put a small value of say 5s in there, the dial command will probably give up before someone answers at the other end. Neither is workable. Asterisk SHOULD be able to distinguish between a TRYING and no response. In the event it gets no TRYING response to a dial command within a specified timeout it should return control and flag an error. If on the other hand it does get a TRYING response (and maybe a RINGING too) it should continue to wait until the 20s has expired. I can't use dynamic DNS (ie putting two A records for a hostname in DNS) because Asterisk reads the extensions.conf on startup and also seems to cache what the host maps to on startup. Subsequent calls to the host always return the same IP address. But... in general... how have people implemented this? Help appreciated! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Dial Failover
Doug, We currently are using Digium TE410P boards directly into each Asterisk server. I've been researching various gateways, up to DS3 capacity, to convert PRI to SIP and then allocate the SIP among multiple Asterisk servers. I've looked at Cisco AS5400 (), Lucent APX 1000 ($$$), and Quintum Tenor CMS ($$). Thanks, Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, December 09, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover Adam, An Audicodes Mediant 2000 gateway with a couple of PRI's. Why? Doug. -Original Message- From: Adam Robins [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover What are you using to terminate the PSTN calls and do the SIP transcoding? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan All, I have an Asterisk system that sends PSTN calls to an OpenSER system to be routed. I have a command like this in my extensions.conf: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) There's actually two OpenSER systems for redundancy. I'm trying to find a way to have Asterisk attempt to route the call to one OpenSER system, and if it's down, fallback to another. Any first thoughts on how to achieve this? I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups are broken. If I issue a series of Dial commands, such as this: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) ... what seems to happen is that when proxy1 is down, Asterisk waits the full 20 seconds before returning control. Also, This 20s includes the time is takes for the other end to answer, so if I put a small value of say 5s in there, the dial command will probably give up before someone answers at the other end. Neither is workable. Asterisk SHOULD be able to distinguish between a TRYING and no response. In the event it gets no TRYING response to a dial command within a specified timeout it should return control and flag an error. If on the other hand it does get a TRYING response (and maybe a RINGING too) it should continue to wait until the 20s has expired. I can't use dynamic DNS (ie putting two A records for a hostname in DNS) because Asterisk reads the extensions.conf on startup and also seems to cache what the host maps to on startup. Subsequent calls to the host always return the same IP address. But... in general... how have people implemented this? Help appreciated! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please
[Asterisk-Users] VoIP Gateway
We are looking for a high density PRI-to-SIP gateway for our call center and IVR applications. The device must take in a channelized DS3 and output SIP g729a to multiple Asterisk servers. We have looked at the Cisco AS5400XM, Lucent APX 1000 and Quintum Tenor CMS (fronted by an Adtran M13). Can anyone out there provide info about their experiences with the Lucent and/or Quintum products service? Does anyone know where I may find performance comparisons? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium TDM Revision I Card
We had a Rev I card that did not work. We sent it back to Digium and had it reflashed back to H. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob LithSent: Friday, November 11, 2005 1:40 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Digium TDM Revision I Card I had a customer have problems with REV I and J cards get snap, crackel pop noise but not on older REV F or H cards. He upgraded to 1.2.0-rc1 and to quote: "Asterisk 1.2.0-rc1was Released on2005-11-08 22:40. as well as zaptel 1.2.0-rc1. (First non Beta version) I compiled it and it works very nicely, without any Snaps,Cracles or Pops, even though zaptel still detects the REV-J as an REV-I." Regards Rob On 11/11/05, Shaun Singh [EMAIL PROTECTED] wrote: Is anyone using version I TDM mothercard? I am currently using 2 revision Hcards and they are working fine. I recently purchased a revision I card from an online vendor which didn't work and the replacement from Digium (anotherrevision I) didn't work either.Shaun Singh, ManagerTravelwave1655 Dufferin Street, Suite 201Toronto, ON M6H 3L9Tel: (416) 652-1212 Ext 101 Fax: (416) 652-7073Website: www.travelwave.ca___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersThe contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GoToIf Regular Expression
I am trying to test whether a callerid number is a valid ten digit
number. I'm a total novice with regular expressions.
I've tried:
exten = s,n,GotoIf($[${CALLERIDNUM} : \d{10,10}]?label)
But CLI gives an error. Can someone please show me what the correct
syntax would be to do this?
Thanks,
Adam
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[Asterisk-Users] Satellite WAN
We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Satellite WAN
Thank you all for your input on this subject. I think I'll pass for now! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Pyeron Sent: Wednesday, November 02, 2005 2:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Satellite WAN On Wed, 2 Nov 2005, Juan Janczuk wrote: Sattellite links aren't cheap, and, the worst of all, you have in a idel condition, 1.4 seconds latency. I know you can get less, our client in the mid-west uses Hughes with under 600ms. But never attempted to do VOIP over it. -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Partner Sr. Manager 7 West 24th Street #100 - - +1 (443) 921-0381 Baltimore, Maryland 21218 - - - -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, purge the message from your system and notify the sender immediately. Any other use of the email by you is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering Machine Detection
I just checked the 1.2 source. It looks like app_AMD is gone. All references to it on the Wiki are also gone. Can someone please tell me why AMD was removed? I am using it in 1.07 for several production applications. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt FlorellSent: Thursday, October 06, 2005 7:13 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Answering Machine Detection I've read about app_amd and asterisk but there doesn't seem to be much info about it out there. Is it called something else? and where do I look for it?MATT--- On 10/5/05, Adam Robins [EMAIL PROTECTED] wrote: It's already built in.AMD.On Wed, 5 Oct 2005, Cory Andrews wrote: Anyone aware if Digium or Sangoma, or possibly a function of Asterisk, supports answering machine detection on an outbound call? I'll post a detector on Mantis tomorrow (honestly!)SteveThe contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering Machine Detection
It's already built in. AMD. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, October 05, 2005 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Answering Machine Detection On Wed, 5 Oct 2005, Cory Andrews wrote: Anyone aware if Digium or Sangoma, or possibly a function of Asterisk, supports answering machine detection on an outbound call? I'll post a detector on Mantis tomorrow (honestly!) Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iax2 trunking wackyness
I have two Asterisk boxes that I thought were trunked, but based on not
seeing the (T) in iax2 show peers, now I'm not sure.
Server 192.168.xxx.1 extensions.conf has:
Exten = _2XXX,1,Dial(IAX2/interoffice:[EMAIL PROTECTED]/${EXTEN})
Server 192.168.xxx.1 iax.conf has:
[general]
trunk=yes
[interoffice]
type=friend
host=dynamic
context=extensions
secret=password
disallow=all
allow=g729
Server 192.168.xxx.2 extensions.conf has:
Exten = _3XXX,1,Dial(IAX2/interoffice:[EMAIL PROTECTED]/${EXTEN})
Server 192.168.xxx.2 iax.conf has:
[general]
trunk=yes
[interoffice]
type=friend
host=dynamic
context=extensions
secret=password
disallow=all
allow=g729
Should I plug in the actual IP addresses instead of host=dynamic? Also,
I do not currently have register statements.
In iax.conf for these.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Wednesday, September 21, 2005 10:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iax2 trunking wackyness
Andrew Kohlsmith wrote:
On Wednesday 21 September 2005 07:27, Clive wrote:
My setup is:
telco-asterisk(voip)-asterisk{ITSP}telco
Are both your asterisk boxes peered to each other? IIRC trunking ONLY
works between peers.
If you do iax2 show peers in the console, it should show a (T) for
trunked connections.
--
Cheers,
Matt Riddell
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RE: [Asterisk-Users] Huge Echo
Does anyone know how to use ztmonitor to set gain on a PRI circuit via a TE410P card, or is it just for FXO? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marek Zachara Sent: Friday, September 09, 2005 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Huge Echo Pull the clone card out of the system and look for the chipset numbers on the card. Go to the chip manufacturers web site and find the specs for that chip set. The specs will likely tell you the chipset was designed for the US 600 ohm impedance telephone network, and if your country's telco specs are different (which I'm very sure they are), through away the clone card. Without proper impedance matching there isn't anything your going to be do to fix the problem. I can check the chip on monday, but local telco impedance requirements are 600 ohms - just like US. I'm thinking about playing around with increasing/decreasing resistance by placing additional resistors in the circut. Messy, but if it could help... What do you think? Adding resistance has nothing at all to with impedance matching. Resistance will impact the DC loop, but not the AC impendance. The AC impedance is a function of how the chipset was designed. AFAIR, the impedance is not a simple factor, but a combination of passive resistance plus reactance - which usually varies within measured frequency range. Therefore channging the device resistance WILL change its impedance. I know the result will not be perfect, but at least i hope for better load match than it is now. I assume the specified impedance is required within PSTN frequencies which will be roughly 100-4khz, right? Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
Should it be in half duplex or full duplex? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Sunday, August 28, 2005 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards Adam Robins wrote: We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. Check if the network card is in half duplex mode. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
Everything is set to autoneg, NICs, switches and router -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julio Arruda Sent: Monday, August 29, 2005 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards Matt Riddell wrote: Adam Robins wrote: Should it be in half duplex or full duplex? Full. AFAIK, depends... If you have your switches doing autonegotiation, you can't disable autoneg in the NIC and hardcode it to do 100/Full-duplex, or you WILL have a duplex mismatch. This is as per the standard. A duplex mismatch is really bad, is in fact worse than having segments doing halfduplex (properly). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Softphone Quality Network Cards
We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. This week we rebuilt the entire LAN with Cisco 2950-EI switches and have employed QoS on the switches and router. Still sounds terrible. What we are now finding is that the network card in the PC may be the key to the problem. A Dell Optiplex P4 2.4GHz 512MB machine with an onboard Intel NIC is bad, while an older Dell Dimension P3 864MHz 128MB machine with onboard 3COM sounds good. Has anyone out there had a similar experience? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
We are using Plantronics H51N headset top with DA55 USB adapter which has DSP built-in. Terrible means garbled, unintelligible, underwater-sounding. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Friday, August 26, 2005 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards Hi! We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. Have you tried a different sound card and/or a USB handset (which includes an external sound card)? And what exactly do you mean with terrible sound? Philipp ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speex QoS
Can anyone out there please tell me what ports Speex uses? I want to set up QoS on switches but I can't seem to find this information anywhere. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Speex QoS
So, then these would be the same ports defined in RTP.conf? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Edwards Sent: Monday, August 08, 2005 8:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Speex QoS speex is a codec. it's not a network protocol or a service. you need to be looking to be providing QOS for RTP data, over which the speex encoded data is sent. cheers, Mark On 8/8/05, Adam Robins [EMAIL PROTECTED] wrote: Can anyone out there please tell me what ports Speex uses? I want to set up QoS on switches but I can't seem to find this information anywhere. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- regards, Mark P. Edwards FWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom phones w/ two lines on different servers
I have configured my phone following your example, but it does not work for me. Can you also please share your sip.cfg settings? Thanks, Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Tuesday, August 02, 2005 3:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom phones w/ two lines on different servers I do it all the time, the phone on my desk has four server registrations. Don't use OVERRIDE or web configurations, do it this way: ?xml version=1.0 standalone=yes? PHONE_CONFIG phone1 preferences voice.codecPref.G711Mu=1 voice.codecPref.G711A= voice.codecPref.G729AB=2 / cfg httpd.cfg.enabled=0 / volume voice.volume.persist.handset=1 voice.volume.persist.headset=1 voice.volume.persist.handsfree=1 / gains voice.gain.rx.analog.ringer=10 voice.gain.rx.analog.handset=10 voice.gain.rx.analog.headset=10 voice.gain.rx.analog.chassis=10 / gains voice.gain.rx.analog.ringer=5 voice.gain.rx.analog.handset=2 voice.gain.rx.analog.headset=2 / call call.donotdisturb.perReg=1 / ringtype se.rt.enabled=1 se.rt.1.ringer=9 se.rt.1.type=ring / microbrowser mb.main.home=http://polycom.mason.home/index.html; / SNTP tcpIpApp.sntp.address=81.169.154.116 tcpIpApp.sntp.daylightSavings.enable=0 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.gmtOffset=-14400 tcpIpApp.sntp.resyncPeriod=86400 / reg reg.1.address=100 reg.1.auth.password=secret reg.1.auth.userId=1001 reg.1.displayName=1001 reg.1.label=1001 reg.1.type=private reg.1.server.1.expires=3600 reg.1.server.1.address=192.168.0.1 reg.1.server.1.expires.lineSeize=30 reg.1.server.1.port=5060 reg.1.server.1.register=1 reg.1.server.1.retryMaxCount=0 reg.1.server.1.retryTimeOut=0 reg.1.server.1.transport=2 call.serverMissedCall.1.enabled=1 msg.mwi.1.callBack=8500 msg.mwi.1.callBackMode=contact msg.mwi.1.subscribe= se.rt.1.ringer=9 reg.2.address=200 reg.2.auth.password=secret reg.2.auth.userId=805 reg.2.displayName=805 reg.2.label=805 reg.2.type=private reg.2.server.1.expires=3600 reg.2.server.1.address=207.44.xxx.xxx reg.2.server.1.expires.lineSeize=30 reg.2.server.1.port=5060 reg.2.server.1.register=1 reg.2.server.1.retryMaxCount=0 reg.2.server.1.retryTimeOut=0 reg.2.server.1.transport=2 msg.mwi.2.callBack=8500 msg.mwi.2.callBackMode=contact msg.mwi.2.subscribe= call.serverMissedCall.2.enabled=1 se.rt.1.ringer=11 reg.3.address=300 reg.3.auth.password=secret reg.3.auth.userId=301 reg.3.displayName= reg.3.label=GT reg.3.type=private reg.3.server.1.expires=3600 reg.3.server.1.address=64.246.xxx.xxx reg.3.server.1.expires.lineSeize=30 reg.3.server.1.port=5060 reg.3.server.1.register=1 reg.3.server.1.retryMaxCount=0 reg.3.server.1.retryTimeOut=0 reg.3.server.1.transport=2 call.serverMissedCall.3.enabled=1 msg.mwi.3.callBack=8500 msg.mwi.3.callBackMode=contact msg.mwi.3.subscribe= se.rt.1.ringer=2 reg.4.address=400 reg.4.auth.password=secret reg.4.auth.userId=1500 reg.4.displayName=MAIN reg.4.label=MAIN reg.4.type=private reg.4.server.1.expires=3600 reg.4.server.1.address=192.168.200.1 reg.4.server.1.expires.lineSeize=30 reg.4.server.1.port=5060 reg.4.server.1.register=1 reg.4.server.1.retryMaxCount=0 reg.4.server.1.retryTimeOut=0 reg.4.server.1.transport=2 call.serverMissedCall.1.enabled=1 msg.mwi.1.callBack=8500 msg.mwi.1.callBackMode=contact msg.mwi.1.subscribe= se.rt.1.ringer=3 / msg msg.bypassInstantMessage=1 / /phone1 /PHONE_CONFIG -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom phones w/ two lines on different servers
This is in the -app.log file: 0804194926|sip |4|00|Registration failed User: 1800, Error Code:403 Forbidden Where '1800' is the extension I am attempting to register. SIP.conf is set up properly, and there is nothing in Asterisk showing a denied registration attempt. Could it be because the second server is on a different subnet across a WAN link? There is no firewall between the phone and the servers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Thursday, August 04, 2005 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom phones w/ two lines on different servers I use the default. Try this. cd /home/PlcmSpIP cat log/YOURMAC-boot.log se what the log file says, also do the same with the YOURMAC-app.log -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom phones w/ two lines on different servers
I have server info there as well. According to the Admin Guide, the info placed in the individual phoneMACADDRESS.cfg file is supposed to override sip.cfg. I'll give it a shot. Thanks. From: [EMAIL PROTECTED] on behalf of Tarpo, Louie Sent: Thu 8/4/2005 5:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom phones w/ two lines on different servers My Polycom 300 is registered on two different servers on two different subnets. It was failing the same way for me as well because we had server information in sip.conf, so it was always going to one server. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Robins Sent: Thursday, August 04, 2005 2:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom phones w/ two lines on different servers This is in the -app.log file: 0804194926|sip |4|00|Registration failed User: 1800, Error Code:403 Forbidden Where '1800' is the extension I am attempting to register. SIP.conf is set up properly, and there is nothing in Asterisk showing a denied registration attempt. Could it be because the second server is on a different subnet across a WAN link? There is no firewall between the phone and the servers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Thursday, August 04, 2005 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom phones w/ two lines on different servers I use the default. Try this. cd /home/PlcmSpIP cat log/YOURMAC-boot.log se what the log file says, also do the same with the YOURMAC-app.log -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Network Troubleshooting Help Needed - Will Pay $$$
Basically, we have a multi-site Asterisk call center application we tried to bring up last week.When the agent places an outbound call ( or takes an inbound call), the agent can hear the customer just fine, but the customer has issues hearing the agent. This does not happen every time and not from the same agent workstation. We have placed sniffers on the Asterisk servers and are seeing UDP checksum errors and packets out of sequence. Each time we think we have ruled out a possible cause, we then contradict ourselves. We have now exhausted our ownlevel of expertise and then some. There is much more to it than this, and I do not expect to resolve this over the mail. I am looking for an expert. Nota small-businessAsterisk configuration person (no offense), but someone who ishands-on withtheinner workings of Asterisk networking. I need someone to work with us on a consulting basis, and travel if necessary. Time is of the essence. Please email me privately if you feel you can assist. Thanks, Adam [EMAIL PROTECTED] The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel update, Asterisk 1.2 janitor projects
Title: [Asterisk-Users] Zaptel update, Asterisk 1.2 janitor projects The Changelog for Zaptel 1.0.9.1 has only one fix listed: -- continue fxo operation after the magical 25 days Could someone please translate this highly technical explanation into something more meaningful?I already spend far too many hours dealing with the "nuances" of Digium hardware. I installed two TDM400P cards in two separate servers last month. Although they continue tolaunch outbound calls, they both mysteriously stopped answering inbound calls until I reboot the systems. I didn't count how many daysthey lasted. I need to know if this zaptel patch addresses this particular issue - because I'm about to toss the cards and order from Sangoma. Thanks From: [EMAIL PROTECTED] on behalf of Russell BryantSent: Mon 7/25/2005 1:44 PMTo: Asterisk Developers Mailing List; asterisk-users@lists.digium.comSubject: [Asterisk-Users] Zaptel update, Asterisk 1.2 janitor projects Greetings!A new version of Zaptel (1.0.9.1) has been released that includes a fixfor fxo modules on tdm cards. If you are using tdm cards, it is veryimportant that you upgrade for your card to work properly.We are hoping to release Asterisk 1.2 very soon and we need your help!If you have some interest in doing some programming, check out thejanitor projects web page:http://dev.asteriskdocs.orgIf you're not a programmer, there is still work to be done. The bugtracker is full of patches that need testing. Hop on IRC and join#asterisk-bugs for help finding things to work on.Thanks!Russell___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Running Asterisk on a Dell PowerEdge 2850 ServerRe: Dell Hardware
Title: [Asterisk-Users] Running Asterisk on a Dell PowerEdge 2850 ServerRe: Dell Hardware It's Digium, not Dell. I have two identical Dell 1850s, each with the allegedly offensive built-in E100 Ethernet ports. I placed a TE410P card in each. One worked great, the other would not modprobe. Upon examination, we discovered that the two TE410P cards had different firmware revisions. Turns out the one with the older version was the one that worked. We sent the second card back and they re-flashed it to the older version. It now works just fine. I had similar problems in a Dell 1750 with TDM400P. The Rev H card worked, but the Rev. I would not. From: [EMAIL PROTECTED] on behalf of M OSent: Sat 7/23/2005 2:34 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Running Asterisk on a Dell PowerEdge 2850 ServerRe: Dell Hardware Hello,Just chiming in here:From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] OnBehalf Of[EMAIL PROTECTED]Sent: Viernes, 22 de Julio de 2005 01:24 p.m.To: Asterisk Users Mailing List - Non-CommercialDiscussionSubject: RE: [Asterisk-Users] Dell Hardware Mmhh nice !! So, why did Digium forbid it :)?If Dell is so bad... why is a Dell 2850 server one ofthe two listed on the compatibility list for ABE?http://www.digium.com/index.php?menu=product_detailcategory=softwareproduct=ABEtab=compatibilityI am running Asterisk on a 100Mbps Pipe on thefollowing:Hardware InformationProcessors 4 -(should be 2)Model Intel(R) Xeon(TM) CPU 3.00GHzChip MHz 2992.81 MHzCache Size 1024 KBSystem Bogomips 23907.52PCI Devices 00:1f.1 IDE interface: Intel Corp.82801EB/ER02:0e.0 RAID bus controller: Dell PowerEdge ExpandableRAID controller 406:07.0 Ethernet controller: Intel Corp. 82541GI/PIGigabit Ethernet Controller07:08.0 Ethernet controller: Intel Corp. 82541GI/PIGigabit Ethernet Controller0b:0d.0 VGA compatible controller: ATI TechnologiesInc Radeon RV100 QY [Radeon 7000/VE]IDE Devices hda: TEAC CD-ROM CD-224ESCSI Devices MegaRAID LD 0 RAID0 69G (Direct-Access)PE/PV 1x6 SCSI BP (Processor)USB Devices Linux 2.4.21-27.0.1.ELsmp ehci-hcd IntelCorp. 82801EB USB2 00:1d.7USB UHCI Root Hub bca0USB UHCI Root Hub bcc0Sincerely,Martin__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto Dial Out
Title: Re: [Asterisk-Users] editing ring time I am using the auto-dial-out feature to play recordings. I create the call files, place them in the outgoing directory and off they go. The problem is that the number I am dialing does not get stored in CDR. One suggestion was to put this number in the callerid field. Problem with that is that the recipient will see their own number, which is unacceptable. I must show a toll-free number. I've tried resetting the callerid in thedialplan context before the CDR is stored. That works great, except if the call goes unanswered, it never makes it into the dialplan logic. I must somehow get this number into CDR, as I need it to match back to a customer activity database. Any suggestions? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer Problem
I do not want to use the default key of '#' for call transfer, because as we all know, it interferes with many IVRs that require # as a termination character. I modified features.conf and added: [featuremap] atxfer = ** The double-star now works great. If I press it while on a call, I go into transfer mode. The problem is that the # still works as well! Shouldn't the atzfer specification turn off the #? Any insight would be appreciated. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music oh hold
I am using
rawplayer:
default =
custom:/var/lib/asterisk/mohmp3/raw,usr/bin/rawplayer
as in: http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it
However, the music is too loud.
Without having to rerecord it, is there a parameter like quietmp3 that can be
used with the above to lower the volume level?
From: [EMAIL PROTECTED] on
behalf of Marcel van Kaam, FoneticaSent: Thu 6/30/2005 3:29
AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] Music oh
hold
Change from default to
manual. I did that and it helped.
Later I changed to
madplay and set that as default. Below my line from
musiconhold.conf:
default =
custom:/usr/share/asterisk/mohmp3/,/usr/bin/madplay --mono -R 8000
--output=raw:-
Marcel
-Original
Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano GrandisSent: donderdag 30 juni 2005
9:05To: Asterisk Users Mailing List -
Non-Commercial DiscussionSubject: R: [Asterisk-Users] Music oh
hold
This is my
musiconhold.conf and my folder:
[EMAIL PROTECTED]:/etc/asterisk# less
musiconhold.conf[classes]default =
quietmp3:/var/lib/asterisk/mohmp3;loud =
mp3:/var/lib/asterisk/mohmp3;random =
mp3:/var/lib/asterisk/mohmp3,-z;unbuffered =
mp3nb:/var/lib/asterisk/mohmp3;quietunbuf =
quietmp3nb:/var/lib/asterisk/mohmp3; Note that the custom mode cannot handle
escaped parameters (specifically embedded spaces);manual =
custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048
--mono -s[EMAIL PROTECTED]:/etc/asterisk# ls
/var/lib/asterisk/mohmp3/fpm-calm-river.mp3 fpm-sunshine.mp3
fpm-world-mix.mp3[EMAIL PROTECTED]:/etc/asterisk#
I think is
ok.
Any ideas
?
Giordano
Da:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Chris StinsonInviato: mercoledì 29 giugno 2005
22.39A: Asterisk Users Mailing
List - Non-Commercial DiscussionOggetto: RE: [Asterisk-Users] Music oh
hold
Does your
default look like this in musiconhold.conf, default =
quietmp3:/var/lib/asterisk/mohmp3
If so, do
you have any music in the directory mohmp3?
-Chris
StinsonNetwork Operations CenterISDN-Net, Inc.615-221-4200
x103[EMAIL PROTECTED]
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano GrandisSent: Wednesday, June 29, 2005 12:35
PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Music oh
hold
Sorry, i
also tried this:
exten
= 6000,1,Answerexten = 6000,2,MusicOnHold(default)
and i got
this result:
*CLI
-- Executing Answer("SIP/2391-8cdd", "") in new stack --
Executing MusicOnHold("SIP/2391-8cdd", "default") in new stackJun 29
19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music
on hold (class 'default') on channel SIP/2391-8cdd == Spawn extension
(local, 6000, 2) exited non-zero on 'SIP/2391-8cdd'
Any ideas
?
Thanks
Giordano
Da:
Giordano Grandis Inviato:
mercoledì 29 giugno 2005 19.27A:
asterisk-users@lists.digium.comOggetto:
Hi, I installed mpg123 v0.59r
without error and defined as defaut folder /var/lib/asterisk/mohmp3. When i set
a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk
1.0.8
*CLI --
Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new
stack -- Called 2391 --
SIP/2391-79a0 is ringing -- Saved useragent "PA168S" for
peer 2319 -- SIP/2391-79a0 answered
SIP/2339-4da6 -- Attempting native bridge of SIP/2339-4da6
and SIP/2391-79a0 -- Started music on hold, class
'default', on SIP/2339-4da6 -- Stopped music on hold on
SIP/2339-4da6 == Spawn extension (local, 2391, 1) exited non-zero on
'SIP/2339-4da6'
Anyone can help me please
?
Thanks
Giordano
The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.
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RE: [Asterisk-Users] Music oh hold
No, I am not using mpg123 at all.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano
GrandisSent: Thursday, June 30, 2005 9:35 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: R:
[Asterisk-Users] Music oh hold
Did u installed mpg123 0.59r ?
Giordano
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Adam
RobinsInviato: giovedì 30 giugno 2005 13.01A: Asterisk
Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List -
Non-Commercial DiscussionOggetto: RE: [Asterisk-Users] Music oh
hold
I am using
rawplayer:
default =
custom:/var/lib/asterisk/mohmp3/raw,usr/bin/rawplayer
as in: http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it
However, the music is too loud.
Without having to rerecord it, is there a parameter like quietmp3 that can be
used with the above to lower the volume level?
From: [EMAIL PROTECTED] on
behalf of Marcel van Kaam, FoneticaSent: Thu 6/30/2005 3:29
AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] Music oh
hold
Change from default to
manual. I did that and it helped.
Later I changed to
madplay and set that as default. Below my line from
musiconhold.conf:
default =
custom:/usr/share/asterisk/mohmp3/,/usr/bin/madplay --mono -R 8000
--output=raw:-
Marcel
-Original
Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano GrandisSent: donderdag 30 juni 2005
9:05To: Asterisk Users Mailing List -
Non-Commercial DiscussionSubject: R: [Asterisk-Users] Music oh
hold
This is my
musiconhold.conf and my folder:
[EMAIL PROTECTED]:/etc/asterisk# less
musiconhold.conf[classes]default =
quietmp3:/var/lib/asterisk/mohmp3;loud =
mp3:/var/lib/asterisk/mohmp3;random =
mp3:/var/lib/asterisk/mohmp3,-z;unbuffered =
mp3nb:/var/lib/asterisk/mohmp3;quietunbuf =
quietmp3nb:/var/lib/asterisk/mohmp3; Note that the custom mode cannot handle
escaped parameters (specifically embedded spaces);manual =
custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048
--mono -s[EMAIL PROTECTED]:/etc/asterisk# ls
/var/lib/asterisk/mohmp3/fpm-calm-river.mp3 fpm-sunshine.mp3
fpm-world-mix.mp3[EMAIL PROTECTED]:/etc/asterisk#
I think is
ok.
Any ideas
?
Giordano
Da:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Chris StinsonInviato: mercoledì 29 giugno 2005
22.39A: Asterisk Users Mailing
List - Non-Commercial DiscussionOggetto: RE: [Asterisk-Users] Music oh
hold
Does your
default look like this in musiconhold.conf, default =
quietmp3:/var/lib/asterisk/mohmp3
If so, do
you have any music in the directory mohmp3?
-Chris
StinsonNetwork Operations CenterISDN-Net, Inc.615-221-4200
x103[EMAIL PROTECTED]
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano GrandisSent: Wednesday, June 29, 2005 12:35
PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Music oh
hold
Sorry, i
also tried this:
exten
= 6000,1,Answerexten = 6000,2,MusicOnHold(default)
and i got
this result:
*CLI
-- Executing Answer("SIP/2391-8cdd", "") in new stack --
Executing MusicOnHold("SIP/2391-8cdd", "default") in new stackJun 29
19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music
on hold (class 'default') on channel SIP/2391-8cdd == Spawn extension
(local, 6000, 2) exited non-zero on 'SIP/2391-8cdd'
Any ideas
?
Thanks
Giordano
Da:
Giordano Grandis Inviato:
mercoledì 29 giugno 2005 19.27A:
asterisk-users@lists.digium.comOggetto:
Hi, I installed mpg123 v0.59r
without error and defined as defaut folder /var/lib/asterisk/mohmp3. When i set
a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk
1.0.8
*CLI --
Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new
stack -- Called 2391 --
SIP/2391-79a0 is ringing -- Saved useragent "PA168S" for
peer 2319 -- SIP/2391-79a0 answered
SIP/2339-4da6 -- Attempting native bridge of SIP/2339-4da6
and SIP/2391-79a0 -- Started music on hold, class
'default', on SIP/2339-4da6 -- Stopped music on hold on
SIP/2339-4da6 == Spawn extension (local, 2391, 1) exited non-zero on
'SIP/2339-4da6'
Anyone can help me please
?
Thanks
Giordano
The contents of this email
message and any attachments are confidential and are intended solely for
addressee. The information may also be legally privileged. This transmission is
sent in trust, for the sole purpose of delivery to the intended recipient. If
you have received this transmission in error, any use, reproduction or
dissemination of this transmission is strictly prohibited. If you are not the
intended recipient, please immediately notify the sender by reply email and
delete this message and its attachments, if any.
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To
RE: [Asterisk-Users] TDM card and voicemail volume
I was able to raise the volume from inaudible to acceptable by increasing the RxGain in zapata.conf by 5db. I'd rather not go the uncomressed wav route, as it will chew up storage in my email system. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, June 27, 2005 11:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM card and voicemail volume I saw some conversation about this in the archives, but nothing definitive. If a call comes in over a CO line via the TDM400P, the Comedian Mail recording volume is so low it's inaudible. Calls coming in via SIP or IAX do not have this problem. Does anyone have any information on this issue? Its still a problem. It seems the greater the distance from the Central Office, the greater the problem (due to the cable loss to the Central Office plus the problem with the TDM card). Part of the problem is there are very few people that understand zaptel, wctdm, drivers, hardware (chipsets) and transmission engineering. Actually, there is only one person and he is now very busy doing other things that are apparently more important. As someone else mentioned, changing to wav format improves the levels a little bit, but its certainly not a fix. There are no known work arounds. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM card and voicemail volume
Hello, I saw some conversation about this in the archives, but nothing definitive. If a call comes in over a CO line via the TDM400P, the Comedian Mail recording volume is so low it's inaudible. Calls coming in via SIP or IAX do not have this problem. Does anyone have any information on this issue? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comedian Mail User Setup Prompts
I have a user who goes into Comedian Mail for the first time and goes thru the initial setup, changes password, records name, etc. Problem is that every time he calls in, it thinks that it's his first time and keeps reprompting him. His password change is reflected in voicemail.conf. Others do not have this problem. Where does Asterisk maintain the first time flag? Any ideas would be appreciated. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P and Dell Poweredge 1750
I installed a new Digium TDM400P in a Dell 1750 server. The system would not recognize the card. I took the FXS modules off of it and put them on another TDM400P card I already had. Old card worked fine with new modules. Old card is Rev. H and new card is Rev. I. Anyone else having any issues with TDM400P rev. I? Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P Channel Group
I installed a TDM400P with 4 FXO modules. Before moving all of my
office phone lines to it, I decided to move only one for testing. I
plugged it into port 4 on the card.
In zaptel.conf I have:
fxsks=1-4
And zapata.conf:
context=incoming
signalling=fxs_ks
busydetect=yes
callprogress=no
musiconhold=default
usecallerid=yes
callerid=asreceived
group=1
channel = 1-4
When I launch an outbound call as ZAP/g1/${EXTEN}, Asterisk goes to
Zap/1 and I hear dead air because there is no line attached to that
port. Shouldn't it be smart enough to go to Zap/4 as the only available
port in the group?
Thanks,
Adam
The contents of this email message and any attachments are confidential and are
intended solely for addressee. The information may also be legally privileged.
This transmission is sent in trust, for the sole purpose of delivery to the
intended recipient. If you have received this transmission in error, any use,
reproduction or dissemination of this transmission is strictly prohibited. If
you are not the intended recipient, please immediately notify the sender by
reply email and delete this message and its attachments, if any.
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RE: [Asterisk-Users] RE: TDM400P Channel Group
I guess that my definition of first available trunk (either forward or
backward) differs from Digium. I would think that the card should know
which ports had an electical signal attached.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Kawakami
Sent: Wednesday, June 22, 2005 12:47 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: TDM400P Channel Group
-Original Message-
Message: 18
snip
When I launch an outbound call as ZAP/g1/${EXTEN}, Asterisk goes to
Zap/1 and I hear dead air because there is no line attached to that
port. Shouldn't it be smart enough to go to Zap/4 as the only available
port in the group?
-you obviously read the wiki enough to know about g1 as a parameter
telling
* to grab the first available line in group one *from the front to the
back*. Look at the wiki again and look for the G1 parameter.
In either case, I don't believe that * would skip over a channel that
didn't have an active line attached to it because that channel is not
seen as offhook (in use). It would still grab the 4th line in your case
and you would get dead air.
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The contents of this email message and any attachments are confidential and are
intended solely for addressee. The information may also be legally privileged.
This transmission is sent in trust, for the sole purpose of delivery to the
intended recipient. If you have received this transmission in error, any use,
reproduction or dissemination of this transmission is strictly prohibited. If
you are not the intended recipient, please immediately notify the sender by
reply email and delete this message and its attachments, if any.
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[Asterisk-Users] Zap POTS Line Problem calling outbound
I have one POTS line going into a TDM400P. Here in Atlanta, we have 10 digit local dialing. I launch a call Zap/1/7705551212 and it goes thru just fine. The next time I try it, without any modifications, I get a Bell recording telling me that I must dial the area code and seven digit number when placing a local call. It's like Asterisk may be starting the dial before the line is ready (I'm guessing). The only thing I could think of was to play with the echotraining parameter. Didn't work. Any ideas would be appreciated. Zaptel.conf: loadzone = us defaultzone=us fxsks=1 Zapata.conf context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400 relaxdtmf=yes rxgain=0.0 txgain=0.0 immediate=no busydetect=yes callprogress=no musiconhold=default usecallerid=yes callerid=asreceived group=1 channel=1 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850
I am having this exact problem today. I have two Dell 1850's running Asterisk 1.07. Both had TDM400P cards running just fine. I replaced the TDM400P in both machines with TE410P. Server One works just fine with just a new modprobe. Server 2 does not even see the card upon reboot. Swapped cards between servers, and the problem stayed with Server 2. Disabled the ee1000 on the Server 2 - still does not see the card. Now recompiling the kernel with ee1000 module, but am very skeptical. Why one, but not the other? Both machines were built by Dell on the same day and have the same OS (Redhat Enterprise 386 Release 3) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Cianfarani Sent: Tuesday, June 21, 2005 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850 Does anyone know what the reason why Dell servers cause so many problems for the digium hardware? Better question any Dell models that don't have any these problems with the digium hardware? Thanks John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoa Sent: Tuesday, June 21, 2005 3:40 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ee1000 Ethernet in Dell 1850 Hmm, i dont think thats the reason they dont recommend the dell server. The problems with the ee1000 kernel module are easily resolved, compile the module into the kernel. Zoa, Andres wrote: Digium's site now lists the Dell 1850 as a potential problem server, as it uses the intel ee1000 Ethernet chipset (as do a majority of servers in the market!). To my knowledge, ALL dell servers with Gigabit interfaces now use the same chipset. Does this mean the Digium cards can't be used in Dell servers unless you disable the onboard ethernet? I don't want to disable the onboard interface, as I use the IPMI management facility for lights-out management. I have a 2850 that doesn't have any audio problems (the reason that I contacted Digium in the first place), so I'm wondering if Digium are simply guessing at problems. Does anyone know anything specific about the supposed incompatibilities with the ee1000 kernel module? I am not sure where you got that chipset reference but all our PowerEdge 1850s come with: Ethernet controller: Intel Corp. 82541GI/PI Gigabit Ethernet Controller ...and they work fine with the TE410. There seems to be an ever-growing list of situations where you can't use the Digium cards. This is a concern to me. ___ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice and Inbound DTMF
I have Broadvoice set up with dtmfmode=inband. All was working just fine. Suddenly today I noticed that if someone calls in to my Asterisk box thru the Broadvoice number, the system no longer recognizes the DTMF tones. I also tried rfc2833 and info. Any ideas? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice and Inbound DTMF
Nevermind. It is now working. Must be Broadvoice. Surprise! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Wednesday, June 15, 2005 9:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Broadvoice and Inbound DTMF I have Broadvoice set up with dtmfmode=inband. All was working just fine. Suddenly today I noticed that if someone calls in to my Asterisk box thru the Broadvoice number, the system no longer recognizes the DTMF tones. I also tried rfc2833 and info. Any ideas? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Windows IAX Softphone
Title: Message Try DIAX. Works just fine! http://www.laser.com/dante/diax/diax.html From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeromy GrimmettSent: Monday, May 23, 2005 12:09 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Windows IAX Softphone Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com [EMAIL PROTECTED]The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dell Poweredge 1850 and Zaptel
If anyone out there is running Asterisk with Zaptel and a TDM400P card on a Dell Poweredge 1850 server, please let me know what OS and kernel version you are running. I keep getting errors when modprobing zaptel and am running out of possibilities, other than motherboard incompatibility. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel on Dell Poweredge 1850 with RH Kernel 2.4.21-15
Hello, We are attempting to install a TDM400P card in a Dell Poweredge 1850 server. We are running Red Hat Linux kernel 2.4.21-15. We can compile zaptel and asterisk without incident. When we try to modprobe zaptel, it produces pages of: /lib/modules/2.4.21-15.EL/misc/zaptel.o: Relocation overflow of type 10 for .rodata.str1.1 /lib/modules/2.4.21-15.EL/misc/zaptel.o: Possibly is module compiled without -mcmodel=kernel! /lib/modules/2.4.21-15.EL/misc/zaptel.o: Relocation overflow of type 10 for .data /lib/modules/2.4.21-15.EL/misc/zaptel.o: Possibly is module compiled without -mcmodel=kernel! /lib/modules/2.4.21-15.EL/misc/zaptel.o: Relocation overflow of type 10 for .bss /lib/modules/2.4.21-15.EL/misc/zaptel.o: Possibly is module compiled without -mcmodel=kernel! /lib/modules/2.4.21-15.EL/misc/zaptel.o: Relocation overflow of type 10 for .bss /lib/modules/2.4.21-15.EL/misc/zaptel.o: Possibly is module compiled without -mcmodel=kernel! /lib/modules/2.4.21-15.EL/misc/zaptel.o: Relocation overflow of type 10 for .bss /lib/modules/2.4.21-15.EL/misc/zaptel.o: Possibly is module compiled without -mcmodel=kernel! /lib/modules/2.4.21-15.EL/misc/zaptel.o: Relocation overflow of type 10 for .bss /lib/modules/2.4.21-15.EL/misc/zaptel.o: insmod /lib/modules/2.4.21-15.EL/misc/zaptel.o failed /lib/modules/2.4.21-15.EL/misc/zaptel.o: insmod zaptel failed Does anyone have this configuration working? Digium is telling us to disable the built-in network cards and also to upgrade to the latest 2.5 kernel. Thanks,Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound ANI DNIS format
Hello, Being totally fed up with the lack of quality and reliability from both VoicePulse and BroadVoice, We are switching to a direct IP connection to Global Crossing. We've installed a local point-to-point T1 into their CO, and they will give/take SIP g729a directly and act as the gateway for us. In setting up the inbound SIP service, they are asking the question, In what format do I want my ANI DNIS presented? They provided examples, such as *ANI*DNIS, etc. Does anyone out there know how Asterisk expects to see this information on inbound calls? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 Technology and VoIP Gateway Primer
Why would you use gateways and PRI's when several of the major carriers (ATT, Global Crossing, etc.) also have products that can interface directly with SIP for the same per minute cost? We have a multisite Asterisk call center application and are routing all calls over private VPN to one central Asterisk location from where we have multiple point-to-point T1's going straight into Global Crossing. They are accepting the traffic as SIP g.729a and are handling the gateway themselves. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Callum McGillivray Sent: Friday, April 29, 2005 1:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer Hi Matt everyone else, We have also been steering toward using a gateway for our large installation. Ours differs from your significantly in as much as our setup will involve 8 apartment buildings located throughout the CBD. Each apartment building will have as many as 600 extensions (rooms) with an Asterisk Server in the comms room in the basement. Incoming and Outgoing calls are going to be trunked from the Asterisk box along a fiber link back to our core exchange, where the calls will be handed off to a gateway machine (Cisco?) which will have an impressively large number of PRI's plugged into the back of it. My (very vague) examination so far tells me that I can use something along the lines of a Cisco AS5400 (a couple of which I have kicking around here in the office). Has anyone had experience in handing off / receiving calls from a Cisco AS5400 with Asterisk ? How is it done ? Matt, is this similar to the idea that you have for your project ? What Cisco hardware have you looked at so far ? How many E1/T1 lines are you going to have terminating on your setup ? Cheers, Callum Matt Roth wrote: Michael, Have you decided which PSTN-VoIP gateway you'll use? Not yet, but our preference is a Cisco gateway. Lucent, Quintum, and AudioCodes also make TDM-VoIP gateways. Prior to purchasing any hardware, our entire layout will be posted to this list in detail for review. Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Deb ian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BYOD provider other than broadvoice
I drop every 3-4 call with VoicePulse Connect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Wednesday, April 20, 2005 6:21 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice Michael Lyszczek wrote: Are there any BYOD providers out that that people have had positive experiences with? I have broadvoice and they suck lately. Anyony have anyone with a good amount of peers and not a lot of downtime? I like voicepulse. They raised their rates recently, but they are still reasonable and I haven't had any problems with them since I started using them back in November. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BYOD provider other than broadvoice
I totally concur. I switched from Broadvoice to VoicePulse because users were complaining about call quality. Now, the quality is good -- when it doesn't drop altogether. What could be worse than touting your new VoIP system to a client and having it drop the call? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, April 21, 2005 12:32 PM To: Trevor Harrison; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice Hmmm... Think I would prefer something harder to get provisioned but that doesn't drop calls. Your users must be forgiving as hell... Mine would show up with pitchforks and torches if calls dropped regularly. They get twitchy if the calls just vary too much in quality... 8) Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Harrison Sent: Thursday, April 21, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice On 4/21/05, Adam Robins [EMAIL PROTECTED] wrote: I drop every 3-4 call with VoicePulse Connect. My users are also reporting occasional dropped calls when dialing via VoicePulse Connect. But I love the ease of use and setup with their service. -Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
