[asterisk-users] Unable to register an endpoint after upgrading from 1.4.2.20 to 1.8.3.1
Hello, I've just upgrade from 1.4.2.20 to 1.8.3.1 and some kind of endpoint aren't able to register. Message is: [Jul 16 01:26:15] NOTICE[25443]: chan_sip.c:23511 handle_request_register: Registration from 'sip:user637801' failed for 'X.X.X.X:5060' - No matching peer found sip.conf [user637801] type=friend context=FROMuser637805 host=dynamic secret=password dtmfmode=rfc2833 disallow=all allow=g729 allow=ulaw allow=alaw nat=yes canreinvite=no call-limit=2 REGISTER sip:Y.Y.Y.Y SIP/2.0. From: sip:user637801;tag=e00028-c0a800fd-13c4-50029-508b5-41d85fb5-508b5. To: sip:user637801. Call-ID: e162cc-c0a800fd-13c4-50029-508b5-18c6cea7-508b5. CSeq: 2 REGISTER. Via: SIP/2.0/UDP 192.168.100.2:42904;branch=z9hG4bK-508b5-13aa048e-5ff4a5b9. Max-Forwards: 70. Expires: 60. Authorization: Digest username=user637801,realm=asterisk,nonce=7bf18d37,uri=sip:Y.Y.Y.Y,response=ce8847cf30a69e1c7735de86a82c3e6e,algorithm=MD5. Contact: sip:987987987@192.168.100.2:42904. Content-Length: 0. SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.100.2:42904;branch=z9hG4bK-508b5-13aa048e-5ff4a5b9;received=X.X.X.X. From: sip:user637801;tag=e00028-c0a800fd-13c4-50029-508b5-41d85fb5-508b5. To: sip:user637801. Call-ID: e162cc-c0a800fd-13c4-50029-508b5-18c6cea7-508b5. CSeq: 2 REGISTER. Server: ASTERISK. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. SIP/2.0 403 Forbidden (Bad auth). Via: SIP/2.0/UDP 192.168.100.2:42904;branch=z9hG4bK-508b5-13aa048e-5ff4a5b9;received=X.X.X.X. From: sip:user637801;tag=e00028-c0a800fd-13c4-50029-508b5-41d85fb5-508b5. To: sip:user637801;tag=as710c9007. Call-ID: e162cc-c0a800fd-13c4-50029-508b5-18c6cea7-508b5. CSeq: 2 REGISTER. Server: ASTERISK. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. I think the problem is that REGISTER packet has: sip:user637801, instead of sip:user637801@Y.Y.Y.Y This works in 1.4 version but not in 1.8; it maybe more restrict? I can't add @Y.Y.Y.Y in end point's configuration; is there any option to put in peer configuration to allow this registration? Thanks Imanol -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Noisy Ring Back Tone with TE205P card
Hi,
If anyone has the same problem, I solved it doing:
genzaptelconf -sdv
It might have been a problema with the card or the module.
Regards
Imanol Pardavila escribió:
Hi,
I stilll continue with the problem but I have noticed something new
that maybe a clue. The noise during the call progress is made by the
appearance of the different lines in the asterisk CLI, I mean, each
line is posted in the CLI generates a noise in the call's signallling
tone. For example, if I try doing a call during a resetinterval
option, which reset all free channels, each line posted in the CLI
generates a burst noise.
Any ideas?
Thanks
Regards
Imanol Pardavila escribió:
Hi,
I am having problems with an Asterisk with a Digium TE205P card. The
issue is that the Ring Back Tone is noisy. I am making modem's calls
and this noise influences on the initial negotiation protocol, so
modems have to recall.
My configuration is:
Asterisk version: Asterisk 1.4.21.2
Linux version: CentOS release 5.2 (Final)
Card: Digium TE205P
##zapata.conf#
; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is
intended
; to be #include-d by /etc/zapata.conf that will include the global
settings
;
[channels]
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1
language=es
context=default
switchtype=euroisdn
pridialplan=unknown
prilocaldialpla=national
signalling=pri_cpe
resetinterval=never
group=1
channel = 1-15,17-31
; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
group=2
channel = 32-46,48-62
##zaptel.conf#
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
hardhdlc=16
# Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
hardhdlc=47
# Global data
loadzone= es
defaultzone = es
##extensions.conf###
exten =999888777,1,Goto(JUMP,s,1)
[JUMP]
exten = s,1,Dial(Zap/R2/6,15,r)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Goto(HANGUP,s,1)
exten = s-NOANSWER,1,Goto(HANGUP,s,1)
exten = s-CHANUNAVAIL,1,Goto(HANGUP,s,1)
exten = s-CONGESTION,1,Goto(HANGUP,s,1)
[HANGUP]
exten = s,1,Hangup
[DID_span_1]
include = default
[DID_span_2]
include = default
I have no idea about where could be the problem. I can't see anything
rare in the logs
Any ideas?
Thanks
Regards
___
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Re: [asterisk-users] Noisy Ring Back Tone with TE205P card
Hi,
I stilll continue with the problem but I have noticed something new that
maybe a clue. The noise during the call progress is made by the
appearance of the different lines in the asterisk CLI, I mean, each line
is posted in the CLI generates a noise in the call's signallling tone.
For example, if I try doing a call during a resetinterval option, which
reset all free channels, each line posted in the CLI generates a burst
noise.
Any ideas?
Thanks
Regards
Imanol Pardavila escribió:
Hi,
I am having problems with an Asterisk with a Digium TE205P card. The
issue is that the Ring Back Tone is noisy. I am making modem's calls
and this noise influences on the initial negotiation protocol, so
modems have to recall.
My configuration is:
Asterisk version: Asterisk 1.4.21.2
Linux version: CentOS release 5.2 (Final)
Card: Digium TE205P
##zapata.conf#
; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is
intended
; to be #include-d by /etc/zapata.conf that will include the global
settings
;
[channels]
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1
language=es
context=default
switchtype=euroisdn
pridialplan=unknown
prilocaldialpla=national
signalling=pri_cpe
resetinterval=never
group=1
channel = 1-15,17-31
; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
group=2
channel = 32-46,48-62
##zaptel.conf#
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
hardhdlc=16
# Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
hardhdlc=47
# Global data
loadzone= es
defaultzone = es
##extensions.conf###
exten =999888777,1,Goto(JUMP,s,1)
[JUMP]
exten = s,1,Dial(Zap/R2/6,15,r)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Goto(HANGUP,s,1)
exten = s-NOANSWER,1,Goto(HANGUP,s,1)
exten = s-CHANUNAVAIL,1,Goto(HANGUP,s,1)
exten = s-CONGESTION,1,Goto(HANGUP,s,1)
[HANGUP]
exten = s,1,Hangup
[DID_span_1]
include = default
[DID_span_2]
include = default
I have no idea about where could be the problem. I can't see anything
rare in the logs
Any ideas?
Thanks
Regards
___
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asterisk-users mailing list
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[asterisk-users] Contact id protocol problem
Hi, I'm using an Asterisk box with zap channel as a gateway between PSTN and an alarm receiver system. The alarm system uses Contact ID protocol. My problem is that the negotiation fails and I think that the problem is that kissoff tone is cut and the transmitter doesn't recognize it. Maybe the asterisk tone duration isn't long enough. I'm thinking about increasing the toneduration value in zapata.conf. or changind DTMF tone frecuency. Does anyone deal with a similar problem? What are the optimal values? Thanks Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Noisy Ring Back Tone with TE205P card
Hi,
I am having problems with an Asterisk with a Digium TE205P card. The
issue is that the Ring Back Tone is noisy. I am making modem's calls and
this noise influences on the initial negotiation protocol, so modems
have to recall.
My configuration is:
Asterisk version: Asterisk 1.4.21.2
Linux version: CentOS release 5.2 (Final)
Card: Digium TE205P
##zapata.conf#
; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;
[channels]
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1
language=es
context=default
switchtype=euroisdn
pridialplan=unknown
prilocaldialpla=national
signalling=pri_cpe
resetinterval=never
group=1
channel = 1-15,17-31
; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
group=2
channel = 32-46,48-62
##zaptel.conf#
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
hardhdlc=16
# Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
hardhdlc=47
# Global data
loadzone= es
defaultzone = es
##extensions.conf###
exten =999888777,1,Goto(JUMP,s,1)
[JUMP]
exten = s,1,Dial(Zap/R2/6,15,r)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Goto(HANGUP,s,1)
exten = s-NOANSWER,1,Goto(HANGUP,s,1)
exten = s-CHANUNAVAIL,1,Goto(HANGUP,s,1)
exten = s-CONGESTION,1,Goto(HANGUP,s,1)
[HANGUP]
exten = s,1,Hangup
[DID_span_1]
include = default
[DID_span_2]
include = default
I have no idea about where could be the problem. I can't see anything
rare in the logs
Any ideas?
Thanks
Regards
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
Hi, The SIP messages flow is this: ### AAA.BBB.CCC.DDD: Asterisk 1 IP address EEE.FFF.GGG.HHH: Asterisk 2 IP address ### REGISTER sip:ast2.domain.comSIP/2.0 Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;rport From: sip:0...@ast2.domain.com;tag=as715628d7 To: sip:0...@ast2.domain.com Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:s...@aaa.bbb.ccc.ddd:19646 Event: registration Content-Length: 0 Using latest REGISTER request as basis request Sending to AAA.BBB.CCC.DDD : 19646 (NAT) Transmitting (NAT) to AAA.BBB.CCC.DDD:19646: SIP/2.0 100 Trying Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;received=AAA.BBB.CCC.DDD;rport=19646 From: sip:0...@ast2.domain.com;tag=as715628d7 To: sip:0...@ast2.domain.com Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:0...@eee.fff.ggg.hhh Content-Length: 0 Transmitting (NAT) to AAA.BBB.CCC.DDD:19646: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;received=AAA.BBB.CCC.DDD;rport=19646 From: sip:0...@ast2.domain.com;tag=as715628d7 To: sip:0...@ast2.domain.com;tag=as5ccb43ac Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7294c1d1 ontent-Length: 0D Asterisk 1 sends an REGISTER without credentials, and Asterisk 2 replies with a 401 message (with Digest algorithm, realm and nonce). I want to configure the Asterisk 1 in order to send REGISTER with credentials. Thanks Regards Grygoriy Dobrovolskyy escribió: Paste your register lines (hide pass) 2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com mailto:imanol.pardav...@ibercom.com I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
