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hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030906/ed1d46cf/attachment.htm

hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee

hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to

hi All, quit new to asterisk, can anybody tell me whether Wildcard E400P and Wildcard E100P support R2 CAS protocol. if they do, what is the value, i should set to 'signalling' parameter in the zapata.conf file? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, can we have arrays in contexts? i tried like this, but didn't work :-( declaration myarray[0]=192.168.3.4 myarray[1]=192.168.3.1 usage myvalue = ${myarray[${myval}]} pls tell a way to do this Thanx a lot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030521/f8b61c89/attachment.htm

hi, This question is about DBPut and DBGet, Can i put about 1000 keys in a single family, (only once for the lifetime) for ex. exten => _X.,5,DBput(family/key1=${val}) ... exten => _X.,5,DBput(family/key1000=${val}) like above and if i later retrieve it, randomely, with inbound calls, will it affect performance? Surajee -------------- next part -------------- An HTML

hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with, 4 port station interface card ,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) <------------------------------>

hi, hi we have G729 codec from Digium, without the G729 codec, we can do the hash transfers to other sip phones fine. but once we are using the G729 codec, the asterisk is not responding to hash transfer, ie, when we press "#" it does not detect it and says "transfer..", is this a problem with G729 codec? (for testing purposes we have bought licenses for 2 chs) this also

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to