I have to linux system one server one client ,Iam trying to connect those system with SIP using XLite .Dialing , connectivity is all happening, but sound is not coming . Is it because i dont have any interface card .
Do i have to install any interface card in my linux sytem ,or call transfer can happen without using it also
If you have a SIP phone “here” and a SIP phone “there”, you don’t need a telephone interface card. You just need a TCP/IP network that connects both ends. If, say, you wanted use an analog line to make PSTN calls from those SIP phones, that’s when you would need an interface card in your Asterisk server.
Then Do i have to get connected with the VOIP Service Provider .Since iam able to get connected with each other (Server & Client) but my voice cannot be transfer to the client side (vice versa)
You are able to call one SIP phone from another (and vice versa) without any interface to the public telephone network. Even if the two phones are on other sides of the world, they just need a TCP/IP transport (i.e. the internet). Of course it’s up to you to configure the Asterisk server to allow each phone to register, and then configure the phones for the Asterisk server. (If you’ve done all this but still can’t make SIP-to-SIP calls, the problem isn’t a lack of PSTN connectivity.)
You would need a VoIP service provider (or phone line interface card) if (when) you want either of those SIP phones to call a number on the “regular” telephone network.
Hope this helps.
Thanks Bobby
That was a really a good help . But still even after the connectivity i am not able to hear voice .When i am dialling from client to server on a linux platform using XLite softphone iam getting error
rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.1.48
192.168.1.48 is a server IP , & its behind the NAT
Can I get any help regarding this
Any number of things could be the root of such a problem. But first you want to make sure both SIP phones are in fact registered to the server. You can do that by connecting to the Asterisk command line (asterisk -r) and typing ‘sip show peers’. That will show you what phones, if any, are connected. If none appear, you know where to begin working.
I recommend talking it out further in the IRC channel #asterisk on Freenode.
You should also pick up the Asterisk book, free in its entirety here:
downloads.oreilly.com/books/9780596510480.pdf
I would also read “Switching to VoIP” also published by O’Reilly.
no, from xlite to another xlite, no card is needed. please check your sound card setting and also if there are any nat between asterisk and xlite.
I have made RTP DEBUG.
When i call from Server to client ,i get this message on to asterisk console
Sent RTP P2P packet to 192.168.1.33:8000 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.48:8000 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.33:8000 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.48:8000 (type 00, len 000160)
Sent RTP P2P packet to 192.168.1.33:8000 (type 00, len 000160)
AND When i call from CLient to Server ,i get this message on to asterisk console
Got RTP packet from 192.168.1.48:8000 (type 13, seq 000001, ts 000000, len 000005)
[May 22 11:29:35] NOTICE[8319]: rtp.c:788 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.1.48
Got RTP packet from 192.168.1.48:8000 (type 13, seq 000002, ts 000000, len 000005)
== Spawn extension (sip, 1000, 1) exited non-zero on 'SIP/2000-094a8660'
I am totally confused for past 2 weeks iam not able to figure out this thing
Hi , this my first post.
I have the same problem, everything seems to be working fine but , i have no sound. I can hear Voicemail menu and messages that i have left to the accounts , but i cannot talk during a call.
This is what i get when makking a call
-- Executing [6004@numberplan-custom-1:1] Dial("SIP/6000-08fd86f0", "SIP/6004") in new stack
-- Called 6004
-- SIP/6004-08fdbfe8 is ringing
-- SIP/6004-08fdbfe8 answered SIP/6000-08fd86f0
-- Packet2Packet bridging SIP/6000-08fd86f0 and SIP/6004-08fdbfe8
== Spawn extension (numberplan-custom-1, 6004, 1) exited non-zero on 'SIP/6000-08fd86f0'
I have created users from GUI which are created in file users.conf
and i have added this lines in extensions.conf
exten => 6000,1,Dial(SIP/6000)
exten => 6003,1,Dial(SIP/6003)
exten => 6004,1,Dial(SIP/6004)
exten => 1001,1,VoiceMailMain
the thing is that before adding that in extentions.conf i got an error no IAX something , which was solved , but the very strange thing is that now at USERS GUI page i have the users i have created and i have also greyed out the users i have added. I cannot edit them from GUI.
Voicemail works fine.
Any help?
Thank you in advance.