[Asterisk-Users] announce hold time issues

Dear frinends, I am useing the asteirsk for ACD features my dobut is this in queues.conf we can give the announce hold time yer or no . If we give how yes how can we say that this hold time dynamically to the queuemembers . are we manually setting any where that this call need to be last only for

[Asterisk-Users] Queues Messages not Playing

Dear friends, Ihave a asterisk-1.0.9 verison with me in redhat linux 9.0 I am trying for ACD I have two agents 1001,1002 and one queue called "queue1" My requirement is like when ever any member try to enter into the queue it should say messages like you are next and also hold time the related

[Asterisk-Users] error related to the native formats

Hello friends, i make a call through queue to the agent when agent lifts the call it gives one side voice and i get this message in the debug chan_sip.c:1880 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) in my sip.conf iam allowing only ulaw can

[Asterisk-Users] Queue + optional URL

Hello list, Can any body say what Exactly optinal URL will be used in Queue. It states like this "The optional URL will be sent to the called party if the channel supportsit" but when we will send it to the called user ? and if we send is there any specific use ?. with regards rk Too

[Asterisk-Users] acd+transfer+asterisk-1.0.7

Hello friends, iam getting problems in transfering the call from one agent to another agent iam useing the asterisk-1.0.7 and sj phone i have one queue called 30 for that one i have two agents 9020 anb 9024 and normal memeber 1001 so when 1001 calls to the queue 30 it goes and connect to

[Asterisk-Users] wrapuptime + agents.conf

hello list, i have problem when i am useing wrapuptime with agents.conf my agents.conf looks like this [agents] autologoff=15 musiconhold = default wrapuptime=5 group=1 agent = 1001,4321,Mark Spencer recordagentcalls=yes my aim is every call needs have wrapuptime of 5000 ms but when

[Asterisk-Users] voice quality in asterisk

hello list , iam using a simple setup as shown below ip device --- ser -- asterisk (astcc) ---pstn gatewsy Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

[Asterisk-Users] voice quality with asterisk

hello list , my set up is like this ip device --ser --- asterisk(astcc) -- pstn gatewsy my asterisk version is 1.0.2 iam using the ser as registration and asterisk aa the prepaid one with the help of the astcc. now my problem is the destination people i.e the pstn line s are listening

[Asterisk-Users] open g723+limiting the out bound calls

Hello freinds, i have two doubts 1) has any body used the open g723 is it working if so where can i find the codec_g723.so / codec_g723.c it s not there in the original link . all other files are there but no codec_g723.c 2) i have scenario like this where the numbers starting with 256

[Asterisk-Users] problme with astcc

hello friends, i have strange problem with astcc here is my model forwarding all calls from ser to asterisk for astcc ser ( sip proxy ) astcc ( * ) ser (out bound sip proxy) then again through * to the outbound ser ( sip proxy) here billing is going on nicely if we use the

[Asterisk-Users] ser+ asterisk

hi list, i want to use the astersik in conjunction with the ser so i followed the instructions provided on the voip-info.org site but when calling from one user to another it gives me problem in the asterisk cli that failed user authentication my aim of doing this is to use the