Re: [Asterisk-Users] Grandstream problem
Paul Hewlett wrote: On Friday 25 November 2005 01:45, Alfie Viechweg wrote: 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: --- Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815 handle_request_register: Registration from '' failed for '10.0.3.21' - Username/auth name mismatch Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[EMAIL PROTECTED]' In the web set up page on the phone, did you make sure that the 'Auth ID' is set to 100 ? Paul It was an installation problem. I used INSTALL_PREFIX variable to place the sample files in a staging area and that added the staging area prefix to all the pathnames in asterisk.conf. Editing asterisk.conf fixed the problem. The Makefile has two (2) staging area variables DESTDIR and INSTALL_PREFIX but is not too clear about the uses and result of them. I used the wrong one I guess. Thanks anyway. -Alfie ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream problem
On Friday 25 November 2005 01:45, Alfie Viechweg wrote: > 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: > > --- > Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815 handle_request_register: > Registration from '' failed for '10.0.3.21' - > Username/auth name mismatch > Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms > Destroying call '[EMAIL PROTECTED]' In the web set up page on the phone, did you make sure that the 'Auth ID' is set to 100 ? Paul -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream problem
Michel Belleau (malaiwah.com) wrote: Hi Alfie. Did you try setting up a "username=100" in your [100] context and a "username=101" in your [101] context? That should do the trick.. Michel Belleau SERVICES INFORMATIQUES MALAIWAH.COM (418) 261-6412 -- http://www.malaiwah.com Alfie Viechweg a écrit : 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: ;tag=aea38200ad3c1539 To: Contact: Call-ID: [EMAIL PROTECTED] 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: ;tag=aea38200ad3c1539 To: ;tag=as248942d8 Call-ID: [EMAIL PROTECTED] CSeq: 10001 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815 handle_request_register: Registration from '' failed for '10.0.3.21' - Username/auth name mismatch Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[EMAIL PROTECTED]' * 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() ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I tried adding username=xxx and that did not solve the problem. What is the 'sip show users' command (using CLI) suppose to show in a properly configured server? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream problem
Hi Alfie. Did you try setting up a "username=100" in your [100] context and a "username=101" in your [101] context? That should do the trick.. Michel Belleau SERVICES INFORMATIQUES MALAIWAH.COM (418) 261-6412 -- http://www.malaiwah.com Alfie Viechweg a écrit : > 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: ;tag=aea38200ad3c1539 > To: > Contact: > Call-ID: [EMAIL PROTECTED] > 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: ;tag=aea38200ad3c1539 > To: ;tag=as248942d8 > Call-ID: [EMAIL PROTECTED] > CSeq: 10001 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Max-Forwards: 70 > Contact: > Content-Length: 0 > > > --- > Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815 > handle_request_register: Registration from '' failed > for '10.0.3.21' - Username/auth name mismatch > Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms > Destroying call '[EMAIL PROTECTED]' > > * 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() > > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Michel Belleau (malaiwah.com) n:Belleau;Michel org:MALAIWAH.COM - Services Informatiques adr;quoted-printable:;;6374, avenue Royale;L'Ange-Gardien;Qu=C3=A9bec;G0A 2K0;Canada email;internet:[EMAIL PROTECTED] tel;work:(418) 261-6412 x-mozilla-html:TRUE url:http://www.malaiwah.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream problem
Does everything work fine now? I am still having
problems with SayUnixTime. Voicemailmain2 works
though. The one simple AGI script I wrote doesn't do anything. Asterisk starts
playing and the grandstream just rings. Both work fine on other channels/sip
phones.
Thanks,
Will
- Original Message -
From:
Wim
Venneman
To: [EMAIL PROTECTED]
Sent: Friday, November 07, 2003 1:46
PM
Subject: Re: [Asterisk-Users] Grandstream
problem
Thanks William,
Works fine now.
Wim
- Original Message -
From:
William
Carlson
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 9:43
PM
Subject: Re: [Asterisk-Users]
Grandstream problem
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: [EMAIL PROTECTED]
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
Re: [Asterisk-Users] Grandstream problem
Thanks William,
Works fine now.
Wim
- Original Message -
From:
William Carlson
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 9:43
PM
Subject: Re: [Asterisk-Users] Grandstream
problem
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: [EMAIL PROTECTED]
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
Re: [Asterisk-Users] Grandstream problem
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: [EMAIL PROTECTED]
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
RE: [Asterisk-Users] Grandstream problem
Look, at the codecs compatibility between the phones and “canreinvite=X” in your sip.conf Ta Senad
