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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U) cseq 2 From NexTone 100 Trying cseq 2 From Asterisk 200 OK cseq 2 From As...

...our local DID's in Seattle 2. comes into our Asterisk server in Los Angeles or Denver 3. is routed by Asterisk for termination back to a different Seattle PSTN ....and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone softswitch in Dallas before ultimately terminating at the desired Seattle PSTN line... What is the resulting affect as it relates to any difference in "user experience" for the caller in Seattle....and what, if any, is the cost difference on our end due to the extra hop? CHEER...

Thank`s all, Chris Ziomkowski wrote: > If you only want to use H.323 with Asterisk, you should configure it as an H.323 gateway. > Why are you trying to set "softswitch"? I was asked by a costumer, because he could not use a asterisk as a softswitch in the Nextone configuration, so I`m looking for the difference in asterisk configs files. Thank`s a lot for you help... Everton Goularth

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't see...

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

Hello, we're in process of testing an interconnection with a trans-european carrier. But the carrier wants the SIP signalling to server 1 and the RTP stream to server 2. How do I configure asterisk to work with that type of installation. It seems they are using NexTone as SIP Signaling and RTP servers. Can someone help me??? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@msnetworks.lu IT Service Provider http://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belva...

...afe_asterisk" just sent me an email saying "Asterisk on bill exited on signal 11. Might want to take a peek.". Looking at the /var/log/asterisk/message doesn't show me anything... This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and it is routing calls from Nextone MSW Softswitch to VPS Softswitch... Any reason why Asterisk died? Thanking in advance... Cheers, Nitesh

...systems when RTP timeouts occur. It appears that asterisk doesn't send a BYE when it decides to terminate a call because of a RTP Timeout - is this a configuration problem? if so what need changing? or is this a bug/feature? if so is there a work around? The setup here is calls come from our Nextone softswitch and connect to remote Asterisk boxes and this all works fine unless the Asterisk box decides there is an RTP timeout - it clears the call down as far as the remote (asterisk) end is concerned but the Nextone never sees the BYE so holds the channel up until a Max-Duration timer clears it...

...ctelnum>@ss.callcentric.com> > i: 18075985-3606475083-968100 at msw2.telengy.net > CSeq: 1 INVITE > Max-Forwards: 8 > m: <sip:f1eb8ab7586b3f2b72742b5e4d43d78d at 204.11.192.161:5060;transport=udp> > Supported: timer > c: application/sdp > l: 350 > > v=0 > o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.161 > s=sip call > c=IN IP4 204.11.192.161 > t=0 0 > m=audio 50960 RTP/AVP 18 0 8 101 > a=fmtp:18 annexb=no > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G72...

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable? Francesco Pellegrini pellegrini@frameweb.it

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Hi all, I have the following setup: UAs ------------SER ------------------------ ASTERISK ---------------------GNUGK --------------- GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out

If anyone is looking at upgrading their Digium hardware, or has an excess of any of these cards, please let me know. I am not interested in non-Digium hardware, but thanks for asking :) E100P E400P TE410P I only need one card, in that preference order. Would also prefer if the card is already in Australia. I'm in Sydney/Australia if that matters to you. Regards, Adam -- Adam Goryachev

Come on you carriers on the list... Give up the dibs what are you using and why? About to sledgehammer these SELECT * FROM GARBAGE WHERE SBC = 'nCite' Don't bother shooting me off Newport Networks stuff... Too pricey -- ==================================================== J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 echo infiltrated.net|sed

...Response) Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1245 retrans_pkt: Hanging up call 20e5feaa-1d3d7ef1 at 192.168.1.2 - no reply to our critical packet. Customer can receive inbound calls without any disconnections, its just when he tries to make outbound calls. All outbound calls are sent to Nextone SoftSwitch and default codec is G729a. Customer has Linksys SPA-2102 - firmware ver 3.3.6 and Asterisk version 1.2.18. Thanking in advance... Cheers, Nitesh

Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _________________________________________________________________ The

...g profiles, but I'm experiencing problems with w2k, NT4/XP just not test because i'm testing on my box at home and when it'll work fine then I'll go to clients from which smaller part has XPs installed on PC. I think if it works on w2k so will be working on XP too. My problem is nextone: Logon to domain from w2k box is OK(NO ERROR). Drives are mapped and everything works fine. Just smb_audit tells that the .Desktop.ini cannot found at directory "." ?!? I'm ignoring it, but when I logout the system begin process of storing profile on Samba server and report that s...

...gt; > Hello, > > we're in process of testing an interconnection with a trans-european > carrier. But the carrier wants the SIP signalling to server 1 and the > RTP stream to server 2. How do I configure asterisk to work with that > type of installation. It seems they are using NexTone as SIP Signaling > and RTP servers. Can someone help me??? > > Regards, > > Marc > -- > CTO Marc Storck > MS Networks SA mstorck@msnetworks.lu > IT Service Provider http://www.msnetworks.lu > 15, route d'Esch...

Hello, I have a question about Linux Advanced Routing..Is it possible to have more than 252 user-defined routing tables? I see the table id is an unsigned char everywhere, is there a reason to have it limited to 255? Can I change it to more than that? Any adverse effects? Thanks! _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl