Re: [Asterisk-Users] call waiting disable in sip
Paul Liew wrote: - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 29, 2003 3:34 AM Subject: Re: [Asterisk-Users] call waiting disable in sip what would happend if all operators are busy? would app_queue exit? would it schedule the call to wait and until the number of them reaches the maxlen ( it is defined in queues.conf) ? Hi Anton, Before I submitted the patch to bugtracker to fix this problem, I tested this for both the Dial and Queue apps, and it works as per other channels, ie when all the queue operators are busy, the calling party will stay in the queue until an agent becomes free. All parameters within the queue.conf apply. The only parameter you need to specify in sip.conf is the incominglimit for this to work. For GS phones, set this to 1. By the way, this is no longer a patch as it has been incorporated into the CVS as of 26/11/03. Let me know if you encounter any problems. Paul I have a problem, when caller is in Queue and the operator is busy answering other call he/she still hears the call waiting signal. I have the latest cvs and incominglimit is set to 1. But here is what * shows when the operator is answering ( that is his phone is busy): UsernameincomingLimit outgoingLimit 107 0 1 0 1 and operator is getting a call waiting tone. Coould I be missing something? here is my sip.conf: [107] type=friend host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info defaultip=172.22.0.137 mailbox=201 ; Mailbox for message waiting indicator callerid=ipphone1 201 callgroup=1 pickupgroup=1 incominglimit=1 outgoinglimit=1 extensions.conf is very simple. it just calls Queue: exten = 101, 1, Queue(phila) may I be missing something in granstream phones? Thanks a lot -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting disable in sip
On Mon, Dec 01, 2003 at 03:33:50PM +0200, Anton Yurchenko wrote: I have a problem, when caller is in Queue and the operator is busy answering other call he/she still hears the call waiting signal. I have the latest cvs and incominglimit is set to 1. But here is what * shows when the operator is answering ( that is his phone is busy): UsernameincomingLimit outgoingLimit 107 0 1 0 1 and operator is getting a call waiting tone. Coould I be missing something? here is my sip.conf: [107] username=107 // this is required for chan_sip.c to find the username. type=friend host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info defaultip=172.22.0.137 mailbox=201 ; Mailbox for message waiting indicator callerid=ipphone1 201 callgroup=1 pickupgroup=1 incominglimit=1 outgoinglimit=1 extensions.conf is very simple. it just calls Queue: exten = 101, 1, Queue(phila) Put the `username` parameter in your stanza of the sip.conf for the device. This is necessary for the incominglimit code to find the device that is making the call. If you look in your `debug` logs you'll probably see that the `user` variable is null. may I be missing something in granstream phones? Thanks a lot -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting disable in sip
Walker Haddock wrote: On Mon, Dec 01, 2003 at 03:33:50PM +0200, Anton Yurchenko wrote: I have a problem, when caller is in Queue and the operator is busy answering other call he/she still hears the call waiting signal. I have the latest cvs and incominglimit is set to 1. But here is what * shows when the operator is answering ( that is his phone is busy): UsernameincomingLimit outgoingLimit 107 0 1 0 1 and operator is getting a call waiting tone. Coould I be missing something? here is my sip.conf: [107] username=107 // this is required for chan_sip.c to find the username. thanks, I think that is working i`ll try that in production environment, tommorow, and`ll report that. BTW right now without specifiing username, and incominglimit set to 1, I after a while see that it shows 1 but the phone is not in use at all. And this phone is stuck in this position until reload. Anyone have seen happen? type=friend host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info defaultip=172.22.0.137 mailbox=201 ; Mailbox for message waiting indicator callerid=ipphone1 201 callgroup=1 pickupgroup=1 incominglimit=1 outgoinglimit=1 extensions.conf is very simple. it just calls Queue: exten = 101, 1, Queue(phila) Put the `username` parameter in your stanza of the sip.conf for the device. This is necessary for the incominglimit code to find the device that is making the call. If you look in your `debug` logs you'll probably see that the `user` variable is null. may I be missing something in granstream phones? Thanks a lot -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call waiting disable in sip
Anton Yurchenko wrote: Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find anything like it in the grandstream docs. Thanks You need to apply * patch found here: http://bugs.digium.com/bug_view_page.php?bug_id=408 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting disable in sip
Patrick wrote: On Fri, 2003-11-28 at 09:14, Anton Yurchenko wrote: Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find anything like it in the grandstream docs. Thanks Anton, In sip.conf play with incominglimit= and outgoinglimit=. Brian has fixed whatever wasn't working in cvs: http://bugs.digium.com/bug_view_page.php?bug_id=408 (thanks Brian!). If you want to use this you will need cvs from 11/26 or more recent. what would happend if all operators are busy? would app_queue exit? would it schedule the call to wait and until the number of them reaches the maxlen ( it is defined in queues.conf) ? Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting disable in sip
- Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 29, 2003 3:34 AM Subject: Re: [Asterisk-Users] call waiting disable in sip what would happend if all operators are busy? would app_queue exit? would it schedule the call to wait and until the number of them reaches the maxlen ( it is defined in queues.conf) ? Hi Anton, Before I submitted the patch to bugtracker to fix this problem, I tested this for both the Dial and Queue apps, and it works as per other channels, ie when all the queue operators are busy, the calling party will stay in the queue until an agent becomes free. All parameters within the queue.conf apply. The only parameter you need to specify in sip.conf is the incominglimit for this to work. For GS phones, set this to 1. By the way, this is no longer a patch as it has been incorporated into the CVS as of 26/11/03. Let me know if you encounter any problems. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
