search for: outgoinglimit

Hi all, I have some peers configured in SIP.CONF file with parameters incominglimit and outgoinglimit set up to 10. By doing that, I expect that this peer will not be allowed to handle more than 10 incoming calls and 10 outgoing calls at the same time. However, since I upgraded to Asterisk 1.2.17, I started to face a problem. Sometimes, calls to those peers are not connected. When I check the logs...

For those that don't read every line of source code here's something I found out today... -------- Deprecated incominglimit and outgoinglimit Incominglimit = number of calls the local extension can originate to Asterisk. Outgoinglimit = number of calls Asterisk will terminate to local extension. End of Life for these commands announced**, please use setgroup and checkgroup, that will also be helpful with cross channels. There is an exa...

Hi, I seem to have a problem with chanisavail and the call limits on sip phones(incoming and outgoing) The problem seems to be that chanisavail when trying create to create channels and hanging them up afterwards screw up the current usage limit on the phones. Example with chanisavail: Phone A calls voicemail (usage now 1) Phone B tries to call Phone A and uses ChanIsAvail in the dialplan.

...n their display, and an audible beep is heard over the phone (regular call waiting). I would like to limit the number of calls sent to each phone to 1 call only; otherwise respond as being busy. I have looked at trying to accomplish this in the sip.conf by using the 'incominglimit' and 'outgoinglimit' parameters, however, the only one that *seems* to work is the 'incominglimit'. This prevents further calls from reaching the phones, rings busy, but does not allow our phones to initiate a 2nd call OR transfer their existing call. The 'outgoinglimit' parameter does not seem to...

Hi folks, As none of the SIP softphones that I tested can disable more than one incoming call, I decided to implement it by software ;-) I'm attaching a patch that does it. To make it work, modify your sip.conf file and include callwaiting=[0|1] at the general section, or for each peer that you wish to control. Please note that I haven't tested it too much, and my source tree is quite

I would like to setup a hunt group, not a group ring, using sip phones. Anyone done this with sip devices? Comments suggestions? I have not had much luck with the outgoinglimit=1, incominglimit=1 stuff that I would need to get busy extinctions to work right, which is why I'm asking on the list.

...XX,1,Dial,SIP/BYEXTENSION@iconnect exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN} Well, it's not working because iconnecthere actually picks up the second call and plays a message. So nothing ever gets to line 2 above. I got quite excited when I saw the sip.conf option: outgoinglimit=1 Which one might expect would prevent 2 concurrent calls on iconnect. But that doesn't seem to have any effect. So I've been wondering if there is another way. I guess I could spawn a system command that goes and looks at the current SIP connections. But that seems a bit desperate, no...

...e at least 13 chars long amaflag- Categorization for CDR records. Choices are default, omit, billing, documentation and of course Null though not stated. >>> canreinvite this looks to be 3 chars not 1 (Null, no, yes) >>> context See name above >>> incominglimit/outgoinglimit these have been depreciated and probably should be removed unless there is come Realtime coding that requires these fields to be present. >>> restrictcid currently 1 char long, should be 3 chars for values (Null, no, yes) >>> pickupgroup Since callgroup was set to 30 I jus...

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 Yurchenko<phila@dg.net.ua> Digital Generation

...fromuser | varchar(50) | YES | | NULL | | | fromdomain | varchar(31) | YES | | NULL | | | host | varchar(31) | | | | | | incominglimit | char(2) | YES | | NULL | | | outgoinglimit | char(2) | YES | | NULL | | | insecure | char(1) | YES | | NULL | | | language | char(2) | YES | | NULL | | | mailbox | varchar(50) | YES | | NULL | | | md5secret...

Let's say you have a 256 Kbps Internet connection and you're using it for voice calls. With mu-law (G.711), each call uses about 80 kbps, so you really can't have more than 3 calls active at one time. Does Asterisk support any kind of Call Admission Control where it would prevent you from originating a call if it would exceed your Internet bandwidth? For example, in this case, ideally,

Hello, Since outgoinglimit is EOL'd, I've implemented SetGroup/GetGroupCount to ensure that SIP clients will only have a single call at any time. Works perfectly for simple calls using Dial(). I'm now struggling to find a way to similarily limit 2nd calls to SIP clients that are Agents, who receive their cal...

...call asterisk still sends them more, resulting in a "beep beep" (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a "single line" user agent, like firefly, still this behaviour does not make any sense to me. I tried using incominglimit and outgoinglimit in my sip.conf, even if they are deprecated: no luck. Here is a sample from my queues.conf, something wrong in my setup maybe ? Tnx for any help! [ingombranti] joinempty = strict maxlen=3 musiconhold = default announce = annuncio-ingombranti strategy = rrmemory servicelevel = 60 timeout = 15 an...

Hello, I am running version 1.4 with realtime support. I've set (for Snom phones 300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the database). - When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem was that when such a phone received a call and did attended transfer it was left "in use" and could not receive new calls. - After seeing reference to similar problem on this lis...

...capitalization... For some unknown reason, we had to standardize on M$ Outlook... *sigh*) [1234] Type=friend Context=whatever Host=dynamic Secret=password1234 Dtmfmode=inband Disallow=all Allow=ulaw [1234b] Type=friend Context=whatever Secret=password1234b Dtmfmode=inband Disallow=all Allow=ulaw Outgoinglimit=1 . . . Rinse, lather, and repeat for each queue you want on a phone, or as many call appearances as you have. Since we have IP600s, and nobody is in more than 5 queues currently, it works well for us. We avoid the call waiting issue using the outgoinglimit=1 directive, as the Asterisk server wi...

...-------+-------------+------------+----------------+--------+--------+----------+-------+----------+------------+--------+-----------+ | uniqueid | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | host | incominglimit | outgoinglimit | insecure | language | mailbox | md5secret | nat | permit | deny | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | allow | disallow | regseconds | ipaddr | auth | +----------+------+-------------+----------+-----------+----------+--------...

...s <2153> host=dynamic ; we have a dynamic IP address ;nat=no ; there is not NAT between phone and Asterisk canreinvite=yes ; allow RTP voice traffic to bypass Asterisk dtmfmode=rfc2833 ; either RFC2833 or INFO for the BudgeTone ;outgoinglimit=1 ; disable callwaiting signal (2nd call to phone) ;incominglimit=1 ; permit only 1 outgoing call at a time mailbox=2153@default ; mailbox 1234 in voicemail context "default" disallow=all ; need to disallow=all before we can use allow= allow=u...

...ver? What do they mean by proxy server? I tried to dial 124 and it just dropped into voicemail... Any ideas? Thanks, Lyle sip conf ;uip200 1 [124] type=friend context=local callerid="Lyle" <124> username=124 secret=******** host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 ;outgoinglimit=1 ;incominglimit=1 mailbox=101 disallow=all ;allow=gsm allow=ulaw allow=alaw ;allow=g723.1 Extensions.conf exten => 124,1,Dial(SIP/124,24,Ttr) exten => 124,2,VoiceMail(u101) exten => 124,3,Hangup

Hi Akk, If this has been discussed/done then apologies be-4-hand. I did not find it in the Wiki or the Archives. Here's the question. We have incoming PRI lines, all on the same main number. An attendant is supposed to handle all incoming calls. Now, let's say I have a multi-line SIP phone. For argument's sake (and to keep it simple) say I only have two lines. We'll call them

...henever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes qualifyfreq=300 insecure=port,invite nat=yes outgoinglimit=4 incominglimit=4 [mypeer](peer) host=10.0.138.226 defaultuser=2155551941 fromuser=2155551941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IP (tos 0x0, ttl 64, id 9242, offset 0, flags [none], proto UDP (17), length 555) 10.0.1.3.5060 > 10.0.138.226.5060: SIP, length: 527...