similar to: asterisk not able to hear calling party ring sound

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

Dear all i need this feature in asterisk whn 2 party calling that time i pickup call and listen conversation of that party spoofing like is it possible in asterisk Rgds satish patel --------------------------------- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. -------------- next part -------------- An HTML attachment was

Dear all I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan I have setup asterisk with avaya system i have 5 avaya system on 5 location i use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when

Dear All I am new in this list right now i am working on asterisk server and deploying asterisk PBX in my organization now i have alread setup Avaya PBX and i want to intergrate my asterisk through mediant 2000 [asterisk]-----[mediant 2k]--------E1-trunk------[Avaya] this is my setup now i want to create dialpan so how to forward call in to existing avaya setup means i have not

Dear all I have setup with mediant 2000 with avaya now i want to install E1/PRI card with asterisk and trunk with E1 with Avaya E1 port so i want to buy E1 card for asterisk so which card is best and cast effective for my setup i want 1 port E1 card so can you suggest me which card is best for my setup and i want QSIG singaling with avaya Regards satish patel

Dear All I have this setup [asterisk]----[mediant2000]-------E1 Trunk----------[Avaya] When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error *CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Dear All I want to integrate asterisk with mediant so anybody have configuration for this setup [asterisk]----------[mediant]------[avaya] this is my setup so what is the basic configuration for this setup --------------------------------- Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. -------------- next part -------------- An

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

Hi, We are in the process of moving from an Avaya Definity to Asterisk for our institution's phone system. I got one feature that the Avaya had, which I have not been able to reproduce with Polycom phones and Asterisk; since this feature seemed so small and useless to me when testing, I kind of ignored it. Now I am getting more "I miss that" requests than I expected. =) On the

Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]-------[Mediant2k]------------[Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task

Hi, Is there a way to send and receive SMS over SIP protocol with Asterisk ? I mean, between two SIP phones like below... SIP_phone "A" (sending sms) ====> Asterisk ========>SIP_phone "B" (receiving sms) ... Is it possible ? If so, how could I do it ? Thanks, Angel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

(REPOST, sorry if you get this more than once.) Hello all, (Not sure if this is more appropriate for user or dev list) Does asterisk have any sort of "standards based" api that can enable an application to do call control on the switch ? For example, if I am developing a call center application using asterisk, I would like to be notified of inbound calls and then be able to route

Hi All, I have Asterisk 11 talking to Avaya over SIP trunk using TLS and SRTP. On incoming calls from Avaya asterisk complains of 'unsupported crypto parameters: UNENCRYPTED_SRTCP' and rejects the call with '488 Not acceptable here' Doesn't Asterisk support UNENCRYPTED_SRTCP as crypto parameters in sdp? FYI SDP looks like this. v=0 o=- 1429194215 1 IN IP4 XX.XX.XX.XX s=-

Dear all I have video confranceing deivice Codian and i want to intergrate asterisk box with codian so voice confrance is possible with codian users means some users have not codian endpoint so thay call join confranceing with SIP PHONE I have configures asterisk and register codian in asterisk now whn i call from asterisk to codian i got IVR and ask me to inter confrance

pstn?????????????????????????????????????????????????????????? pstn asterisk???????????????? link between???????????????? avaya pbx both systems tied together by 2 pri's both have trunks out to the pstn want to get rid of the avaya pstn trunk and send thru my asterisk box avaya still has inbound calls on this trunk until late november (at&t is dragging their feet doing the porting - 8

Dear all I have configure asterisk with avaya so now i have configure 11 for trunk line to goes on avaya system but now i want to replace it with 0 means I press ' 0 ' it will convert my digit in 11 automaticaly is there any dialplan to do this ?? Regards satish patel --------------------------------- Need a vacation? Get great deals to amazing places on

Dear all I have recently install TE120P Digium E1 card now everything is fine and working i have connect my asterisk with avaya but when anybody transfer call from avaya i got this error on my asterisk consol [Aug 27 14:46:50] WARNING[19527]: app_dial.c:741 wait_for_answer: Unable to forward voice or dtmf -- Hungup 'Zap/32-1' I m waiting for your reply Satish Patel