search for: limitonpeer

Hi, An observation on this feature, which I may have completely misunderstood, so flame away if I am being dumb :) Looking at the code, setting "limitonpeers=yes" causes all user and peer calls to be ref-counted as if they are peer calls (assuming a user and peer of the same name exist). A side-effect of this is that an incoming call seems to have its call-limit evaluated based on the peer's, rather than the user's settimg, unless no call...

...ndi/secret context=internal [voipprovider] type=friend host=voipprovider.web dtmfmode=rfc2833 insecure=port,invite disallow=all allow=g729 context=external [300] type=peer callerid=300 username=300 secret=secret host=dynamic context=internal mailbox=300 at default notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 [301] type=peer callerid=301 username=301 secret=secret host=dynamic context=internal mailbox=301 at default notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://l...

...9;t offer calls if the user is in a private call etc. We have customers that require both BLF and call waiting at the same time. We are running Asterisk 1.8.11-cert7 I've made the following additions to sip.conf [general]: callcounter=yes counteronpeer=yes (undocumented? Supposed to replace limitonpeers?) (old relevant values, unchanged) allowsubscribe=yes subscribecontext=blf notifyringing=yes notifyhold=yes limitonpeers=yes I also tried may other suggestions I've found like placing the hints in the same context as the extensions and removing subscribecontext. Is there something I'm...

I am trying to get the presence/hints/BLF working along with Realtime SIP but I never get any "busy" notification. core show hints always shows the realtime sip user as idle. I have tried setting call-limit to various values, including 1 but nothing seems to help. I have tried limitonpeers both yes and no. Anybody got any other ideas? I do know the hinting is working as I can "hint" a Zap channel and it works fine. Daniel

...sip.conf [general] bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw tos=0x68 notifyringing=yes notifyhold=yes limitonpeers=yes [120] type=friend secret=secret record_out=Adhoc record_in=Adhoc qualify=yes port=5060 pickupgroup= nat=yes mailbox=120 at default host=dynamic dtmfmode=inband disallow= dial=SIP/120 context=from-internal canreinvite=no callgroup=...

...a module reload of the app_queue.so all the members turn to state "Not in use" and the queue works has expected. - when the members, of the queue, do a sip registration the state all so alters to "Not in use" and the queue works fine. - I have added in the sip.conf "limitonpeer = yes" and "call-limit=10" for a better state update of the members. Question: Is there a simple way of solving my problem without having to do a module reload of the app_queue.so or having a small sip re-registration period? thanks in advance. Alex

...nfo=ring3) exten => _1XX,2,Dial(SIP/${EXTEN},20,Tt) exten => _1XX,3,VoiceMail(${EXTEN}@default,u) exten => _1XX,104,VoiceMail(${EXTEN}@default,b) sip.conf [general] allowsubscribe=yes ;subscribecontext = default notifyringing=no notifyhold=yes ;limitonpeers=yes [100] type=peer context=demo callerid=Back Office <100> username=100 secret=(Private) host=dynamic nat=no qualify=yes canreinvite=no dtmfmode=rfc2833 call-limit=5 mailbox=100 at default disallow=all allow=ulaw allow=alaw ;allow=g723.1 allow=g729 ;callingpres=allowed_passed_screen notify...

...l-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. In sip.conf I have : limitonpeer = yes In my realtime sip_buddies DB I have a column "call-limit" which has a value of '4' for all the sip peers. Still I get the above message... 2nd situation : I should be possible to transfer a call by pressing # followed by the extension, but it does not work. Although...

Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11

2008/11/4 Igor Zamocky <asterix at ponozky.sk> > > Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers > ? > I didn't. Now I did and it's working the way I wanted. Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and SIPPEER but limitonpeers is much more concise. Thanks a lot. > > > Hi, > > > In this thread > http://lists.digium.com/pip...

...nter, for inbound calls I setup a Queue in queues.conf and their respective Agents in agents.conf, but when an Agent is calling out and a call is coming from PSTN the call is send to that agents which have a call in progress. How I can fix this in order to have only one call at a time. I think in limitonpeer and call-limit but the documentation says that the sip user can have 2 calls with this parameters (1 for inbound and 1 for Outbound and this is the behavior I don't want). Thanks in advance. Ricardo MR

...hen a SIP user is busy, he still receive calls from asterisk. I've tried to setup the call-limit preference to 1, but with this kind of configuration the user can't transfer calls, as the system block the 2nd call generated to do the transfer. I've also tried to set the user as friend, limitonpeers = yes and call-limit =1. In that case the work-around works but only when the user is the receiver of the call that makes him busy. If the user is the caller, he still receive a second call. So, isn't there any method to limit the call available for a user to 1 but granting him the possibili...

...unt: ======================================= [intphones](!) type=friend qualify=yes host=dynamic callgroup=1 pickupgroup=1 dtmfmode=sip [1](intphones) context=IntPhones username=1 secret=1234 amaflags=documentation accountcode=11 subscribecontext=IntPhones callerid="phone 11" <11> limitonpeers=yes call-limit=100 [2](intphones) context=IntPhones username=2 secret=1234 amaflags=documentation accountcode=12 subscribecontext=IntPhones callerid="phone 12" <12> limitonpeers=yes call-limit=100 ======================================= and on extensions.conf my dial lines are li...

...e could dial out without any problem in the same network. After we had downgrade to 1.2.32 everything works fine again on these phones. my question is, had there been a big change in sip.conf or codec handling which cause this problem, cause i used the same sip.conf just adding notifyringing=yes, limitonpeers=yes and allowsubscribe=yes. Here is my sip.conf with one client: [general] context=incoming realm=softpbx bindport=5060 bindaddr=0.0.0.0 srvlookup=yes useclientcode=yes defaultexpirey=3600 vmexten=voicemail disallow=all allow=alaw allow=ulaw allow=gsm ;qualify=no ;canreinvite=no musicclass=d...

...result my CLi is on fire with 'busy' notices, because it's trying to ring an agent even when they are on a call. If I remove the state_interface, it shows them as 'busy' in the queue, and doesn't ring them. Let's see, what else did I forget? Other details: sip.conf: limitonpeers=yes and call-limit=5 on each SIP device queue.conf: ringinuse=no Anything else I should look for? Thanks! -Rob

...could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic busylevel=1 limitonpeers=yes call-limit=1 when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ? [May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage limit of 1 -- Couldn't cal...

...e SIP trunk that support multiple DID. Only the trunk is documented in sip.conf (called DID is taken from the sip-header in real time). I would like to limit the number of simultaneous calls on each DID. Is there a way to achieve this ? My understanding is that the SIP configuration parameter "limitonpeers" will limit at the trunk level, right ? Thanks in advance Patrick

...seem to be dropped, just seems to go to voicemail. The agents are also mentioning they do not receive the 30 second wrapuptime I have specified in queues.conf. We're using polycom 501 phones and I'm adding agents to the queues using Addqueuemember(). I believe I have the call limits and limitonpeer settings right in sip.conf. The only difference between the two queues is one has a higher weight. Any suggestions would be greatly appreciated. [our-support-queue] musicclass = default strategy = leastrecent timeout = 12 retry = 15 wrapuptime=30 weight=0 autopause=yes maxlen=0 joinempty=stri...

...look at the queue a few sec after the Agent is marked 'in use' which wasn't the case with IAX iirc Agent are defined using a Local channel, but we used the '/n' flag to passthru the status: Agent/136 (Local/136 at queues/n) .. As for the SIP peer definition i have the limitonpeer=yes in the general section and all peers are templated based on this: [poste](!) type=friend host=dynamic qualify=yes call-limit=6 Is their something more in can do to avoid the warning ?

...f course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device <21168> canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain...