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...CONF ; SIP Configuration for Asterisk [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = from-sip register => mylogin:mypassword@sipauth.deltathree.com/2905 [iconnect] type=friend username=mylogin password=mypassword host=natrelay.deltathree.com #EXTENSIONS.conf [general] static=yes writeprotect=no [globals] [intern] include => iconnecthere [iconnecthere] ;Outbound calls to PSTN numbers via iconnecthere.com's SIP service exten => _61NXXNXXXXXX,1,Dial,SIP/${EXTEN-1}@iconnect exten => _61NXXNXXXXXX,2,Congestio...

...st_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames Here is my sip.conf [general] port=5060 context=default allow=g723.1 maxexpirey=180 defaultexpirey=160 ;Connect to iconnect register=1510xxxxxx:xxxx@natrelay.deltathree.com/1510xxxxxx [iconnect] type=friend secret=xxxx username=xxxxxxx host=natrelay.deltathree.com dtmfmode=inband canreinvite=no context=vobb-in allow=g723.1 Can someone be able to debug this ? If I make the codec to g729, call not even get through. * complains that it can't bridge...

...a slightly different question. At the current CVS can the * SIP client register with a SIP provider (such as ICH) when the * box itself is behind a NAT. I have an ICH account which I can configure fine on an ATA 186 using their "NAT instructions" . Can the SIP client on * be set to use natrelay.deltathree.com as the gateway and/or outbound proxy ? Any info appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030506/faf17adf/attachment.htm

hi i got iconnect to work, works pretty well now except calls sometimes (more often than not) hang up after a couple of minutes.. heres a bit of the debuging Record-Route: <sip:61892142222@213.137.73.178:5060;maddr=213.137.73.176> From: sip:61892142222@natrelay.deltathree.com;tag=3281050172-73809 To: "JUSTIN XLITE" <sip:2001@61.95.68.84>;tag=as09766a78 Call-ID: 5df848b10eea268512db62e41181dc3d@61.95.68.84 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2001@61.95.68.84> Content-Lengt...

...my asterisk setup is talking with iconnecthere. My sip.conf is setup to disallow=all, allow=gsm, allow=ulaw, allow=alaw, however it doesn't make a difference if I connect with either codec. Here is my iconnecthere section with user/pass removed: [iconnect] type=friend secret= username= host=natrelay.deltathree.com dtmfmode=inband canreinvite=no All this seems pretty standard. What settings or debug information should I look at to fix this issue. Has anyone else encountered this problem before. I'd be happy to supply more information about my setup but since I'm not sure where the p...

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

...ried). n= my Iconnect phone number p=Iconnect password u=Iconnect User id sip.conf [general] port = 5060 ;bindaddr = 0.0.0.0 ;bindaddr = 212.199.7.106 context = from-sip ;callerid=No CallID ;register=nnnnnnnnnnn:pppp@sipauth.deltathree.com/ nnnnnnnnnnn ;register= uuuuuuuu:pppp @natrelay.deltathree.com/phone ;register=uuuuuuuu:pppp@sipauth.deltathree.com/phone [iconnect] type=friend secret=pppp username=uuuuuuuu host=sipauth.deltathree.com ;host=natrelay.deltathree.com ;fromdomain=xxx.xxx.xxx.xxx ;dtmfmode=inband nat=no ;nat=yes canreinvite=no disallow=all allow=gs...

...ing used to access a SIP provider such as ICH? I basically want to set the * SIP client so that it registers with ICH for incoming calls. The * system is behind double NAT. An ATA 186 works fine (apart from the normal ICH blitches.) using both the gateway/proxy and outgoing proxy settings (set to natrelay.deltathree.com). Does *'s SIP channel have equivalent support ? Thanks for any info. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030515/8f7fe83b/attachment.htm

Hello all, I have a 7960 registered to asterisk. I am trying to use iconnect as my sip provider. When I send an invite to delta-three, I get the normal INVITE - 407 - INVITE exchange. The problem is, asterisk is sending the second invite using the 'dialed number' from the 7960 as the username, and not my 'username' configured in sip.conf. I believe that digest authentication

Hi All, I can only seem to get iconnecthere working with incoming calls intermittently. One minute it seems to work, and the next it doesn't. I am not aware of anything being changed in the config files. Outgoing calls work ok all the time. The Asterisk box is behind NAT so that does complicate things slightly. However, the Iconnecthere PCPhone client software works perfectly for

...thout any problem but Inbound calls are very intermittent. It seemed to work for a week or so but over the past week 99% of inbound calls are dropped to ICH voicemail. Would moving the Asterisk box to a public IP resolve the problem or is it just an ICH/Asterisk problem? I am registering against natrelay.deltathree.com. asterisk -vvvc shows an "unable to determine sequence number from ''" message. The box is running the CVS build from yesterday. Thanks, Nathan.

...nnection to IConnecthere.com working. I didn't have a register command in sip.conf at first, so I believe that is why it was not working. However, I can't seem to get the register command correct, it just keeps timing out. Below is what I have: register=<username>:<password>@natrelay.deltathree.com I know that there is supposed to be /<extension> at the end of that command. But I can't figure out what the <extension> is supposed to be. I have tried my username and the phone number that they have given me, but it keeps timing out on trying to register. Than...

...ed the call ends and the Asterisk console shows an "Unsupported Media" error as follow: Got SIP response 415 "Unsupported Media" back from 213.137.73.147 My only allowed codecs are alaw and ulaw. My sip.conf looks like: [iconnect] type=friend secret=xxxx username=yyyyyyy host=natrelay.deltathree.com dtmfmode=inband [2000] type=friend username=2000 secret=xxxxxxxx host=dynamic context=from-sip mailbox=2000 qualify=no dtmfmode=inband disallow=all allow=ulaw allow=alaw callgroup=1 pickupgroup=1 Suggestions, please? -------------- next part -------------- An HTML attachment was s...

...) for direct dial to iconnect I have clear voice transmission. I have no way of explaining this. My asterisk sip.conf [general] port = 5060 bindaddr = 192.168.255.33 disallow=all allow=ulaw allow=alaw context=bogon-calls [iconnect] context=from-sip type=peer secret=******* username=35205*** host=natrelay.deltathree.com dtmf=rfc2833 callerid="Me" <35205***> canreinvite=no nat=yes [2000] type=friend username=2000 secret=***** host=dynamic defaultip=192.168.255.54 context=from-sip mailbox=2000@local dtmfmode=info callerid="Me" <2000> My extensions.conf [general] stat...

Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040808/ecc99c4a/attachment.htm

...use to provide customer support. And ideas will be gladly accepted. Now for the sample: Sip.conf: [general] port=5060 bindaddr=0.0.0.0 context=from-sip callerid=NO CallID [iconnect] ;sip for iconnect type=friend username=12345678 secret=1234 host=natrelay.deltathree.com [1000] ;my sip extension type=friend secret=1234 iauth=md5 nat=yes host=dynamic reinvite=no canreinvite=no qualify=1000 dtmfmode=rfc2883 callerid="Tomer Shoval" <1000> disallow=all allow=gsm context=default Extensions.con...

I am going to have to find a fix for this problem or I'm going to have to quit using iconnect. About one call in 10 or so, iconnect's gateway gives me an error (console output appended below). So upon receiving the error, which as a 4XX error means, "Fatal," asterisk gives up and drops the call. But not iconnect!! The phone at the other end starts ringing, and rings

Hi is there any ritual sacrifice a newbie has to perform to be welcome on this list? I am new to this whole PBX thing in general and Asterisk in particular. I had hoped that the community on this list would welcome a newbie like myself and help me with some answers to my stupid questions, but somehow it seems to me that nobody likes to respond to somebody who appears to be a complete