search for: limitonpeers

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://li...

...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=...

hello all, I have asterisk 1.4.26 working on CentOS 5.3 and I have the following problem: - when I restart asterisk all the members of the queue are Invalid. - when I make a call to one of the members, of the queue, and then check the state, it turns to "Not in use" for the called phone, and the queue works fine for that member after. - after doing a module reload of the

...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 notifyr...

Hello list, asterisk 1.4.30 2 situations in which call-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

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/pipe...

Hi to all, I need some help, I have an Asterisk Server in a small call center, 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

...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 possibilit...

...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 lik...

...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=de...

...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 call...

...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

Hi all, We have an 8 agent support desk setup with 2 call queues running Asterisk 1.4.5. Every so often agents will receive a call from the queue that only rings once not allowing them time to answer. The call doesn't 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

Hi, I'm a little surprised, up until 1.4.22 my agents where using an IAX channel to ZoIPer Softphone, however since after the upgrade to .22 we experienced a problem with hangup failure between zoiper and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i made them switch to SIP instead. Weird thing is that the 'Not In Use' warning message keep showing

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in