Hi,
I installed Asterisk an all works fine exept for Grandstream.
When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can
make a conversation. = ok
When I call to a softphone with a Grandstream I can pich up the call with the
softphone but the Grandstream keeps ringing like on the other site you
didn't pick up the phone.(even if you do so)
It's the same when I call between two Grandstream phone's. Call from
phone1 to phone 2, I pick up phone2 and afther 3 seconds I get congestion tone
from both phone's.
Info from command *CLI>
-- Executing Dial("SIP/phone2-a030a", "sip/phone1") in new
stack
-- Called phone1
-- SIP/phone1-663a is ringing
-- SIP/phone1-663a answered SIP/phone2-a030a
-- Attempting native bridge of SIP/phone2-a030a and SIP/phone1-663a
== Spawn extension (sip, 1,1) exited non-zero on 'SIP/phone2-a030a'
and I get congestion
Can anyone give me a direction to solve my problem?
Thanks in advance,
Wim
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Look, at the codecs compatibility between the phones and "canreinvite=X" in your sip.conf Ta Senad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031106/3fb6571d/attachment.htm
try
disallow=all
allow=ulaw
under the general section of sip.conf
that half fixes it for me calls between phones work but talking to asterisk has
some problems.
----- Original Message -----
From: Wim Venneman
To: asterisk-users@lists.digium.com
Sent: Thursday, November 06, 2003 2:29 PM
Subject: [Asterisk-Users] Grandstream problem
Hi,
I installed Asterisk an all works fine exept for Grandstream.
When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I
can make a conversation. = ok
When I call to a softphone with a Grandstream I can pich up the call with the
softphone but the Grandstream keeps ringing like on the other site you
didn't pick up the phone.(even if you do so)
It's the same when I call between two Grandstream phone's. Call from
phone1 to phone 2, I pick up phone2 and afther 3 seconds I get congestion tone
from both phone's.
Info from command *CLI>
-- Executing Dial("SIP/phone2-a030a", "sip/phone1") in new
stack
-- Called phone1
-- SIP/phone1-663a is ringing
-- SIP/phone1-663a answered SIP/phone2-a030a
-- Attempting native bridge of SIP/phone2-a030a and SIP/phone1-663a
== Spawn extension (sip, 1,1) exited non-zero on 'SIP/phone2-a030a'
and I get congestion
Can anyone give me a direction to solve my problem?
Thanks in advance,
Wim
-------------- next part --------------
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Can some on help me find the problem here please: I'm using asterisk 1.2.0 with Grandstream GXP-2000 This is the debugging output from asterisk: <-- SIP read from 10.0.3.21:5060: REGISTER sip:10.0.3.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.3.21;branch=z9hG4bK5c77f205e9f991de From: <sip:100@10.0.3.1>;tag=aea38200ad3c1539 To: <sip:100@10.0.3.1> Contact: <sip:100@10.0.3.21> Call-ID: ea87fe4398c81b7c@10.0.3.21 CSeq: 10001 REGISTER Expires: 3600 User-Agent: Grandstream GXP2000 1.0.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 --- (12 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 10.0.3.21 : 5060 (non-NAT) Transmitting (no NAT) to 10.0.3.21:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 10.0.3.21;branch=z9hG4bK5c77f205e9f991de;received=10.0.3.21 From: <sip:100@10.0.3.1>;tag=aea38200ad3c1539 To: <sip:100@10.0.3.1>;tag=as248942d8 Call-ID: ea87fe4398c81b7c@10.0.3.21 CSeq: 10001 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: <sip:100@10.0.3.1> Content-Length: 0 --- Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815 handle_request_register: Registration from '<sip:100@10.0.3.1>' failed for '10.0.3.21' - Username/auth name mismatch Scheduling destruction of call 'ea87fe4398c81b7c@10.0.3.21' in 15000 ms Destroying call 'ea87fe4398c81b7c@10.0.3.21' ***************** This is the relevant parts of my sip.conf: [100] type=friend secret=test qualify=yes nat=no host=dynamic canreinvite=no context=internal [101] type=friend secret=test qualify=yes nat=no host=dynamic canreinvite=no context=internal ************ This is the relevant part of my extensions.conf: [internal] exten => 100,1,Dial(SIP/100) exten => 101,1,Dial(SIP/101) exten => 611,1,Echo()