RE: [Asterisk-Users] SIP registration
I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
Rémi Letot wrote: Olle E. Johansson [EMAIL PROTECTED] writes: couic I realized the same and started a process to collect a lot of that information and build a knowledge base on http://www.voip-forum.org/ Everyone is right, it should be http://www.voip-info.org I confused with my attempt at blogging at http://www.voip-forum.com , being a late night and a tired mind in Sweden. don't now and simply add What's a pyroflax? on it. Someone will notice and explain what a pyroflax is... A what ? :-) Google ;-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration
Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration
Thanks, my phone has the next sip setting. Can you help me with correct parameters with the below sip.conf? SIP Server Settings * Server Address: (IP or FQDN) * Port: * Domain Name: * Send Registration Request: (true or false) Gateway Settings Dial Plan: Transport: (UDP tor TCP ) Phone Number: CallerID Name: Port: AEC: (On or OFF) User Name: Password: Thanks for all srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jan Janak Enviado el: viernes, 19 de septiembre de 2003 8:59 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration
Thanks, my phone has the next sip setting. Can you help me with correct parameters with the below sip.conf? SIP Server Settings * Server Address: (IP or FQDN) * Port: * Domain Name: * Send Registration Request: (true or false) Gateway Settings Dial Plan: Transport: (UDP tor TCP ) Phone Number: CallerID Name: Port: AEC ON: (On or OFF) User Name: Password: Thanks for all srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jan Janak Enviado el: viernes, 19 de septiembre de 2003 8:59 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration
No, it is not something you can fix by tweaking the configuration files, you should complain to the authors of the user agent. Anyway, it is a minor problem and I guess that most implementations can overcome it, but you should at least report it to the authors. Jan. On 19-09 09:17, Sergio Serrano Revuelto wrote: Thanks, my phone has the next sip setting. Can you help me with correct parameters with the below sip.conf? SIP Server Settings * Server Address: (IP or FQDN) * Port: * Domain Name: * Send Registration Request: (true or false) Gateway Settings Dial Plan: Transport: (UDP tor TCP ) Phone Number: CallerID Name: Port: AEC: (On or OFF) User Name: Password: Thanks for all srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jan Janak Enviado el: viernes, 19 de septiembre de 2003 8:59 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
[Asterisk-Users] AGI problem
Hi.
I have the next configuration... I dial from my analog phone in the
TDM400P to extension 102, and the second agi works about 1 out of 10
times, the other nine it gives me these error on the asterisk console:
-- Starting simple switch on 'Zap/2-1'
-- Executing Macro(Zap/2-1, receivecall) in new stack
-- Executing AGI(Zap/2-1, receivecall.tcl) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/receivecall.tcl
-- AGI Script receivecall.tcl completed, returning 0
-- Executing Macro(Zap/2-1, followme|s|92624663) in new stack
-- Executing AGI(Zap/2-1, followme.tcl) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/followme.tcl
== Spawn extension (macro-followme, s, 1) exited non-zero on 'Zap/2-1'
in macro 'followme'
== Spawn extension (macro-receivecall, s, 2) exited non-zero on
'Zap/2-1' in macro 'receivecall'
== Spawn extension (home, 102, 1) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
and these in /var/log/asterisk/messages:
Sep 19 00:31:08 WARNING[458770]: File app_agi.c, Line 1277 (run_agi): No
channel, no fd?
Any idea on what might be wrong ? I have tested these way of doing a
follow me routine on 3 different asterisk boxes, with different setups
and i got the same result on all of them. The AGI's are suppoused to do
a more complex task, like search in a database for a followme
enabled/disable and the number to wich forward the call, etc. but, just
for testing purposes, these is giving the same error.
Thank's.
===
1) 1 X100P.
2) 1 TDM400P with 1 fxs.
-
zaptel.conf
-
fxsks=1
fxoks=2
loadzone = us
defaultzone=us
---
zapata.conf
---
[channels]
usecallerid=yes
hidecallerid=no
musiconhold=default
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=no
rxgain=0.1
txgain=0.0
group=1
immediate=no
callerid=asreceived
context=bell
signalling=fxs_ks
channel=1
musiconhold=default
callerid=Telefono de Estudio100
context=home
signalling=fxo_ks
channel=2
extensions.conf
[general]
NContexto = home
NExten = s
NPrioridad = 1
NMacro = none
NPar1 = s
Npar2 = s
Extension = s
static=yes
writeprotect=no
[dialout]
exten = _9X.,1,StripMSD,1
exten = _X.,2,Dial,Zap/1/BYEXTENSION
exten = 500,1,Playback(demo-abouttotry); Let them know what's going on
exten = 500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call
the Asterisk demo
exten = 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site
[bell]
include = mailboxes
exten = s,1,setmusiconhold,default
exten = s,2,responsetimeout,20
exten = s,3,Dial,Zap/2SIP/angelSIP/danny|15|tTm
exten = s,4,BackGround,thanks
exten = s,104,BackGround,thanksbusy
exten = i,1,hangup
[mailboxes]
exten = 100,1,Voicemail,100
exten = 100,2,Hangup
exten = 101,1,Voicemail,101
exten = 101,2,Voicemail,
exten = 102,1,Voicemail,102
exten = 102,2,Voicemail,
exten = 103,1,Voicemail,103
exten = 103,2,Voicemail,
[home]
ignorepat = 9
include = dialout
include = parkedcalls
exten = 004,1,Echo()
exten = 004,2,Hangup()
exten = 005,1,Meetme()
exten = 005,2,Hangup()
exten = 008,1,VoicemailMain
exten = 008,2,Hangup
exten = 007,1,Record,thanks:gsm
exten = 007,2,Wait,1
exten = 007,3,Playback,thanks
exten = 009,1,MusicOnHold(random)
exten = 100,1,setmusiconhold,default
exten = 100,2,Dial,Zap/2|20|tTm
exten = 100,3,Congestion
exten = 100,4,Hangup()
exten = 101,1,Dial,SIP/danny|20|tTm
exten = 101,2,Congestion
exten = 102,1,Macro(receivecall)
[macro-receivecall]
exten = s,1,agi(receivecall.tcl)
exten = s,2,macro(${NMacro}|${NPar1}|${NPar2})
[macro-followme]
exten = s,1,agi(followme.tcl)
exten = s,2,noop
exten = s,3,goto(${NContexto}|${NExten}|1)
-
receivecall.tcl
-
#!/usr/bin/tclsh
source /var/lib/asterisk/agi-bin/agilib.tcl
parametros
puts stdout SET VARIABLE Extension $agi(extension)
gets stdin linea
puts stdout SET VARIABLE NMacro followme
gets stdin linea
puts stdout SET VARIABLE NPar1 $agi(extension)
gets stdin linea
puts stdout SET VARIABLE NPar2 92624663
gets stdin linea
exit 0
--
followme.tcl
--
#!/usr/bin/tclsh
source /var/lib/asterisk/agi-bin/agilib.tcl
parametros
puts stdout VERBOSE \mensaje 1\ 1
gets stdin linea
puts stdout SET VARIABLE NContexto dialout
gets stdin linea
puts stdout VERBOSE \mensaje 2\ 1
gets stdin linea
puts stdout SET VARIABLE NExten 96388210
gets stdin linea
puts stdout VERBOSE \mensaje 3\ 1
gets stdin linea
exit 0
--
agilib.tcl
--
proc parametros {} {
global agi
set linea [gets stdin]
while { [string length $linea] 0 } {
set subindice [string first : $linea]
if { $subindice 0 } {
set agi([string range $linea 4
Re: [Asterisk-Users] Need help with H.323
- Several people have told me chan_capi can work as a gatekeeper, so there should be no use for gnugk or any others. I have yet to find where this is hidden. FWICS, this is all commented out (ast_h323.cpp line 722). Is this right or have I overseen anything? i can confirm that chan_capi will not work as a gatekeeper under any circumstances! somebody is obviously trying to confuse you.;-) er. right I meant chan_h323 :P Any more questions? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Radio for Music on Hold?
You can't legally do this. At least no here in the US. AFAIK, you can do this in norway, and probably in the rest of europe. I know several doing it on norway already. roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P?? [Static problem]
-= On Thu, 18 Sep 2003 23:32:07 -0400, Sean Rodger [EMAIL PROTECTED] said: There seems to be a problem with the (not my) TDM400P hardware. Here is the origional problem: Sometimes there is a clean dialtone. Sometimes there is a loud crackling sound over the dialtone. I just recently installed a new TDM400P and I am facing similar problems: Freshmaker version: 63 Freshmaker passed register test Module 0: Initialized Module 1: Not installed Module 2: Not installed Module 3: Not installed Found a Wildcard FXS: Wildcard TDM400P REV E (4 modules) I have not had the device reset or fail as far as the software is concerned, but it does have an enormous amount of static on the line. A phone connected to the TDM400P (only one port) will hear a loud crackling over the dialtone. The reverse (microphone) audio is even worse -- almost totally drowned out by noise. I can post a recorded sample if it would be helpful, but it sounds like crispy white noise. Asterisk can not decode the touch-tones, of course. Stranger yet, the card seems to have gotten worse over the last four days. When it was first installed, it was fine. Then a little static, then a lot of static, and now it's useless. Temperatures have been stable, and no other hardware changes explain it. I have tried connecting an 8-pin RJ45 cable with RJ11 adapter, with no improvement. Power alarm on module N, resetting! I do not get this error. I have only one phone attached. I've tried multiple known-good phones. I have the power connector attached (since it would not work without it). The machine (Intel P-III) has been stable for over two years and has four other PCI cards in it. The X101P Wildcard FXO works fine. I'm looking for clues as well, especially regarding the degenerate behavior. -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * website needs a place for
This should be a list to come find support and not get jumped on! The * website should instruct where to find information better... I totally agree. Let's patch up a list of things poorly documented, and start there. Just append your comments to my list: - chan_h323 is not documented at all. It has been said that it's able to run as a gatekeeper, but not how etc etc. - AGI is sparsely documented. An explaination of what actually happens in FAQ or perhaps a HOWTO will let people into the game a lot faster, and people won't have to ask so many (silly?) questions on the list or IRC channel. ...and probably a lot more ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Radio for Music on Hold?
On Fri, 2003-09-19 at 10:43, Roy Sigurd Karlsbakk wrote AFAIK, you can do this in norway, and probably in the rest of europe. I know several doing it on norway already. I just rang back a number that had left no message so I had no idea who would answer. I found it totally confusing because at first it sounded like the old crossed line situation. Then I realised he was talking about the traffic situation. When the switchboard picked up I realised it was the autoroute management company who had called. No legal problems for them because it's their own radio, but still confusing. At least musak is musak. My 0.02¤ worth. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP error messages
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 18 September 2003 18:12, marrandy wrote: I'm seeing this at the console. NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' What's this all about ? Pretty straight forward. A SIP phone at '192.168.1.70' failed registration at your Asterisk box at '192.168.1.1'. Try sip debug at your CLI, and you'll see similar messages as the ones I described in my SIP registration thread. (I still can't make the damned thing work) - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/asxv2TEAILET3McRAovYAJ9GAFOo1ANJekQwhUgIYEhZMaJKtwCgk3os vCeIOKqfjV9XmPzjWL4gfFY= =7Y3l -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 18 September 2003 19:04, Hielke Christian Braun wrote: try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. It didn't. And now something else is weird. Asterisk fails sending audio to my SIP phone. Found this in my logs: Sep 19 11:08:52 WARNING[950291]: File channel.c, Line 1819 (ast_channel_make_compatible): No path to translate from SIP/sc.sc.sc.sc-de54( 4) to H323/ip$hc.hc.hc.hc:1244/14060(8) Sep 19 11:08:58 WARNING[147466]: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call [hex]@ as.as.as.as for seqno 102 (Request) Sep 19 11:09:04 WARNING[147466]: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call [hex]@ as.as.as.as for seqno 102 (Request) What on earth is this? Codec? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/as2r2TEAILET3McRAtIaAJ9Hpa3k/a7giiB62pwn7qw17jck/ACeJLdH fzoRqSVrEMfgAfzE5BOogoU= =N4hn -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP registration between *'s
Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =usuario1:pass1@public_ip_2 In* two sip.conf [usuario1] type=friendusername=usuario1 secret=pass1host=public_ip_1dtmfmode=inband Logs in * are the followings In * one logs: Sip read: SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK488fe503;received=public_ip_1From: sip:usuario1@public_ip_2;tag=as504a35d0To: sip:usuario1@public_ip_2;tag=as2a0e47ceCall-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1CSeq: 103 REGISTERUser-Agent: Asterisk PBXContact: sip:usuario1@public_ip_2Content-Length: 0 9 headers, 0 lines11 headers, 0 linesReliably Transmitting:REGISTER sip:public_ip_2SIP/2.0Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2From: sip:usuario1@public_ip_2;tag=as4f879ac7To: sip:usuario1@public_ip_2Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1CSeq: 104 REGISTERUser-Agent: Asterisk PBXExpires: 120Contact: sip:s@public_ip_1Event: registrationContent-length: 0 (no NAT) topublic_ip_2:5060Sip read: SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2;received=public_ip_1From: sip:usuario1@public_ip_2;tag=as4f879ac7To: sip:usuario1@public_ip_2;tag=as13445743Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1CSeq: 104 REGISTERUser-Agent: Asterisk PBXContact: sip:usuario1@public_ip_2Content-Length: 0 In * two logs: NOTICE[81926]: File chan_sip.c, Line 4816 (handle_request): Registration from 'sip:usuario1@public_ip_2' failed for 'public_ip_1' Sip read:REGISTER sip:public_ip_2SIP/2.0Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106From: sip:usuario1@public_ip_2;tag=as35957f60To: sip:usuario1@public_ip_2Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1CSeq: 119 REGISTERUser-Agent: Asterisk PBXExpires: 120Contact: sip:s@public_ip_1Event: registrationContent-length: 0 11 headers, 0 linesUsing latest request as basis requestSending to public_ip_1: 5060 (NAT)Transmitting (NAT):SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106;received=public_ip_1From: sip:usuario1@public_ip_2;tag=as35957f60To: sip:usuario1@public_ip_2;tag=as1538b8a6Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1CSeq: 119 REGISTERUser-Agent: Asterisk PBXContact: sip:usuario1@public_ip_2Content-Length: 0 Any idea to fix the problem Any special configuration in sip.conf Thanks a lot.
Re: [Asterisk-Users] SIP registration between *'s
Why? Use IAX2, it is s much better... J On Fri, 19 Sep 2003 11:54:23 +0200 Xisco [EMAIL PROTECTED] wrote: Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =usuario1:pass1@public_ip_2 In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=public_ip_1 dtmfmode=inband Logs in * are the followings In * one logs: Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK488fe503;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as504a35d0 To: sip:usuario1@public_ip_2;tag=as2a0e47ce Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 103 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 (no NAT) topublic_ip_2:5060 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2;tag=as13445743 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 In * two logs: NOTICE[81926]: File chan_sip.c, Line 4816 (handle_request): Registration from 'sip:usuario1@public_ip_2' failed for 'public_ip_1' Sip read: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 11 headers, 0 lines Using latest request as basis request Sending to public_ip_1: 5060 (NAT) Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2;tag=as1538b8a6 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 Any idea to fix the problem Any special configuration in sip.conf Thanks a lot. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration between *'s
That's true if always there to connect two asterisk servers, but I'm doing some proves in order to connect one asterisk server with another SIP server. That's the matter. - Original Message - From: Jamie Carl [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 19, 2003 12:12 PM Subject: Re: [Asterisk-Users] SIP registration between *'s Why? Use IAX2, it is s much better... J On Fri, 19 Sep 2003 11:54:23 +0200 Xisco [EMAIL PROTECTED] wrote: Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =usuario1:pass1@public_ip_2 In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=public_ip_1 dtmfmode=inband Logs in * are the followings In * one logs: Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK488fe503;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as504a35d0 To: sip:usuario1@public_ip_2;tag=as2a0e47ce Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 103 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 (no NAT) topublic_ip_2:5060 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2;tag=as13445743 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 In * two logs: NOTICE[81926]: File chan_sip.c, Line 4816 (handle_request): Registration from 'sip:usuario1@public_ip_2' failed for 'public_ip_1' Sip read: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 11 headers, 0 lines Using latest request as basis request Sending to public_ip_1: 5060 (NAT) Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2;tag=as1538b8a6 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 Any idea to fix the problem Any special configuration in sip.conf Thanks a lot. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * website needs a place for
Your statement: ''Also a reminder to those who know far more than I, You too started someplace and someone answered your questions and you learned''. Very well said. Almost always, bad and irritable manners are symptoms of deep trauma in one's life. A little tolerance goes a long way. Cheers, Abdul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Flood Sent: 19 September 2003 05:59 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * website needs a place for Hello! This should be a list to come find support and not get jumped on! The * website should instruct where to find information better. Often times the first response to trying to learn something is to ASK a question. I too, first found the archive list tonight. I've been on this list reading since February. Better documentation is the key and since this is a product being developed daily keeping up with the documentation is difficult. It's the new people coming in which keep this idea alive as we, who have been around tell them. What do people see when they read list mail? I see PJ trying to help and John B. who BTW, is also a VOIP reseller, jumping on people who are not changing subject lines. Education and documentation is key to making a product succeed. Possibly a * web page re-design would better educate new people coming into this list so they conform to the lists standards. Also a reminder to those who know far more than I, You too started someplace and someone answered your questions and you learned. Please, lets be considerate of others. Possibly an automated daily message could be sent to the list reminding people to change the subject line or provide a link to the archives... Helping people succeed with * helps everyone who has an interest. Bill Flood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codec probs wit g723.1
Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog --- IAX -INTERNET-IAX---OH323GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codec probs wit g723.1
You would have to buy a g723.1 license which would bust every users budget :) g723.1 is a prpriatory codec and there is no legal implementation for asterisk. On Friday 19 September 2003 1:11 pm, Thomas Haeger wrote: Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog --- IAX -INTERNET-IAX---OH323GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Radio for Music on Hold?
On Thu, 18 Sep 2003 13:21:54 -0700, Paul Crick wrote Tell your client that some callers put on hold may know about the above and radio on hold would make the company look at best ignorent. I read something somewhere.. can't remember where.. some PBX buyer's guide maybe? ANYWAY.. point is.. it sounds bad to the callers.. and you never know what they're hearing.. dodgy music, a DJ going off on one, throwing a fit, an advert for a competitor or something else inappropriate.. Come on people! Fork out $50 for a discman and another few bucks for some royalty free library music and have that on hold instead.. You're in control, you know what your callers are listening to, and you're also legal :-) Oh yeah.. we're talking Asterisk.. the physical connection to an external source is what sparked this whole thread off.. sorry, my bad - I forgot.. ok, forget the discman, fork out for the music, rip it to MP3 and use the built in MOH solution? Or.. are we still talking about the MOH being the output of the radio station that's actually being called, that's using Asterisk as its PBX? Hmm, what do you think about about creating a fake extension (like s, t, fax etc.) called, say, hold that would be called every time moh is played now? to get the old behaviour you'd do: exten=hold,1,MusicOnHold (or sth) and you'd get the required flexibility for just about anything. examples off the top of my head follow: * dial a sip extension which streams an .asf using some proprietary/windows/etc. software * some agi plays you nice music while mixing in some real time generated info (you've been on hold for $time. if you're pissed off already, dial $phone and complain ;)) * well, the top of my head seems to end here but i'm sure you'll find more creative uses :) cheers, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ringing
Does asterisk know when each ring comes in or just the first ring, ie so the cadence can be worked out? say over two rings? Robb Martin Pycko wrote: The X100P together with asterisk does not support the distinctive ringing detection on the line. Asterisk however can generate the distinctive ring over FXS ports. regards Martin I just installed two new x100p cards in the last two weeks; one used on a residential line, the second on a business line (CO Centrex). The business line rings with a Long and a Short within the same time period as a normal ring, which is essentially distinctive ringing. In many US pbx's (and obviously this CO Centrex) the ring is intended to represent a call arriving from an outside line (as opposed to another CO Centrex line). Asterisk failed to recognize the callerid as it believed the callerid data arrived after the second ring (when it fact, it was arriving exactly where it was suppose to by pure telephone standards). I opened a problem with Digium late last week. One of their techs logged into this system, tested with real calls, and observed the problem. They made a source code change in chan_zap.c (and possibly others) and now callerid works fine with that distinctive ring. Since I don't have another copy of the cvs that was in use at the time, I don't know what they changed. I've asked multiple times, but never get a response from the support folks. Therefore, I'm not sure if they fixed a real bug or if they brute-forced this system to look for callerid elsewhere. (And, now I don't know what's going to happen if I apply a current cvs update either.) Since this exact distinctive ring is used by a large number of pbx and telco systems, if * does not handle this properly, Digium is going to either get one hell of a lot of calls or * is going to have to change to properly handle it. Bottom line: the * cvs from about Sep 13 looked for the callerid right after the first ring stopped, and was not based on telephone standards (which are timer based). Based on recent google searches, it would appear Mark was working on some sort of cadence algorithum in early 2002, but I've not found any recent reference that would suggest distinctive ringing is actually supported in current source in any form whatsoever. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
Olle E. Johansson [EMAIL PROTECTED] writes: couic don't now and simply add What's a pyroflax? on it. Someone will notice and explain what a pyroflax is... A what ? :-) Google ;-) No way, even google is moot on that word. I guess you'll have to explain :-) couic -- Rémi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Radio for Music on Hold?
On Thu, Sep 18, 2003 at 01:21:54PM -0700, Paul Crick wrote: Come on people! Fork out $50 for a discman and another few bucks for some royalty free library music and have that on hold instead.. You're in control, you know what your callers are listening to, and you're also legal Why go to all this trouble and expense? - skip the Discman and just rip the royaltyfree CD and save the mp3's on the hard drive. (Check the license to make sure you are allowed to do this). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
Look at all the time you are wasting flaming people. just ignore these questions and get off the high horse. Do you maintain this list? If not then you have no say whatsoever. - Original Message - From: Steve Creel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 18, 2003 6:09 PM Subject: Re: [Asterisk-Users] Grandstream Source? I am NOT a VoIP guru. I am NOT an Asterisk guru. I am NOT a telephony guru. Take that as a disclaimer for the information below, as well as to say that the best learning comes from reading anything you can get your hands on. The idea of post any question to the mailing list works well with 10 people. It scales horribly. Reading through the archives, you will see the same questions asked (and answered) over and over. At _some_ point, it's okay to say I've answered it 15 times, YOU can go look it up on YOUR time. Besides, I'd rather spend 3 hours looking for the answer than just ask my question, because I hate looking like an idiot. This isn't a flame, nor a sarcastic, snide response. I don't want to complain about people asking what is a if I've never made an attempt to answer that question for someone. On Thu, 18 Sep 2003, PJ Welsh wrote: I have to defend us newbies on this. This environment does not facilitate sequential knowledge building! Based on my entry to Asterisk, I should have already known T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get the idea (still trying to figure out skinny...cisco something, I know). Heck, I'm struggling to get a grip on what and how to use/confiure SIP for linux and keep my hair. A T1 is technology used to deliver digital data from one device to another. Most of us are familiar with data T1s - 1.544mbps. When used for voice, they can be PRI (primary rate interface) or Channelized T1. A PRI has 23 voice channels and a bearer channel. The Channelized T1 has 24 voice channels. Depending on the specific application, one may be better suited than another (or depending on the price). There are many other technical characteristics about a T1, but know we've established what it is. An E1 is used for the same purposes as a T1. Which one is it depends on your geographic location - T1 in US, Canada, and Japan (according to a telecom dictionary on the shelf here, sorry if misinformed). Other parts of the world use E1. VoIP refers to the high-level use of an IP network (or IP equipment) to deliver telephone service. Sometimes this means telephone calls from a software app on one machine to another software app. It could mean a call from one physical analog phone to another that was connected by way of an IP network. It could refer to an off-premise extension of your desk phone to home. SIP is session initiated protocol. There are two parts to VoIP protocols - the call setup and the audio stream. All of the audio is handled similarly with most protocols. The difference is usually in call setup. You can use SIP to call from one phone to another directly, without a callmanager, gatekeeper, or any other VoIP equipment. SIP allows IP addresses to be entered and called directly. SIP seems to be best for single-line extensions, I want to call my brother in _ , and for most consumer-grade VoIP for home use. The biggest user experience thing I can think to mention about SIP is that dialing _usually_ (excluding early dial) works like a cellphone - dial number press send. Skinny (or SCCP used interchangably) is Cisco's Skinny Client Control Protocol. It is a proprietary protocol that Cisco uses in their Call Manager system. The Cisco phones use SCCP to talk to the server (yes, like how a SIP phone would use SIP to talk to another phone, or to a SIP server). Because Cisco is Cisco, there is a certain demand to use their devices. To accomodate this, they have offered SIP firmware to load on some of their phones. However, the SIP firmware does not offer all of the features of the firmware for SCCP. Some of this is protocol limitations, some is because they didn't include it. Asterisk's support for SCCP is beginning to be functional (no disrespect to those who have put tons of time in on it already - beginning in that it's beginning to be offered, not beginning to be worked on). FreeWorld is Free World Dialup, or FWD. Their website, www.freeworlddialup.com, says the following: Free World Dialup (FWD) allows you to make free phone calls over the Internet using a 'regular' telephone or a computer program. Free World Dialup does not directly provide access to the traditional telephone networks or cellular networks. FWD members can only call other FWD members and customers of IP-based service providers who have a business relationship with FWD. If you are interested in learning about VoIP and would like to setup your own personal PBX, give Asterisk a try. H.232 is a typo, the protocol is
[Asterisk-Users] ringing tone on analog Zap channel question
Hi all, can somebody explain me why i can't hear a ringing tone (alerting) if i'am going to connect to my destination end point? Is it basically so that i have to configure like: exten = xxx,1,Dial,ChanTec/number|timout|r Is it really nessesary to use the r option everytime if i want to indicate a ringing tone? This suggest a wrong call flow for the user ... Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail2 crashing on replay
Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt': Found Sheriff*CLI Disconnected from Asterisk server -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail2 crashing on replay
Ill need a backtrace. Mark On Fri, 19 Sep 2003, Dave Cotton wrote: Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt': Found Sheriff*CLI Disconnected from Asterisk server -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail2 crashing on replay
Yep Dave same here. It segfaults just as the digit playback starts. This is true even without tz= options set. Holds true with 'make clean' 'make update' 'make' 'make install'. For those that need voicemail, beware. :) Regards, --- Gavin -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED] Sent: Friday, September 19, 2003 9:36 AM To: Asterisk List Subject: [Asterisk-Users] Voicemail2 crashing on replay Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt': Found Sheriff*CLI Disconnected from Asterisk server -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTel registration rejected
Has anybody had a problem registering their IAXtel account? I just signed up for an account and followed the documentation on iaxtel.org and my registration is always rejected. When I type iax show registry, I get the following output: Host UsernamePerceived Refresh State 12.37.165.130:5036Unregistered 60 Rejected (I'm not sure how that will look, but the important thing is that my registration is rejected) I have double checked my password, and it is correct. Is there anything else I should check? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTel registration rejected
Has anybody had a problem registering their IAXtel account? My account is working fine using the following in iax.conf: register = username:[EMAIL PROTECTED] towards the bottom of the [general] section. (I didn't test indial as of this morning to actually validate, but it was working prior.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTel registration rejected
I have that line in my iax.conf --- Rich Adamson [EMAIL PROTECTED] wrote: Has anybody had a problem registering their IAXtel account? My account is working fine using the following in iax.conf: register = username:[EMAIL PROTECTED] towards the bottom of the [general] section. (I didn't test indial as of this morning to actually validate, but it was working prior.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: hangup problem Brazil
- Original Message - From: iPfone Telefonia IP To: [EMAIL PROTECTED] Sent: Friday, September 19, 2003 11:27 AM Subject: hangup problem Brazil Hi all!I´m setting up an asterisk box here in brazil, asterisk don´t hangup afterthe caller disconects...it goes to voice mail etc.. Somebody have the sameproblem?I received that advice from digium support but it dont works:Edit the file "dsp.c" which is in your asterisk source. At the top ofthe file find "#define DEFAULT_THRESHOLD 1024" and change the 1024 to128. Find the "#define BUSY_MIN 75" and change the 75 to 65. Find"#define BUSY_MAX 1100" and change the 1100 to 200. Save the file. Thendelete the file "dsp.o" and then do a "make install". Then reload themodules and start asterisk.When i put callprogress=yes in the conf file the sistem don´t answer thecalls any more, like another postings here.Busydetect=yes dont makes any diference, dont works to...Regards for allMiklos
[Asterisk-Users] Identify call router? How?
I have a machine in my office, it is labelled Cable and Wireless, and on the back it says, SMarT-1 I have searched the web, and no joy. It connects to a PC via a serial cable, has anyone heard of such a device? -- * Not everyone is touched by an Angel Those that are, never forget the experience * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No sound on PSTN -- */PRI
Hi all, i tried to make a call from public pstn in our */E100P. Config is following: exten = _X.,1,Playback(testgsm) But what i hear is one dtmf tone and then nothing... Any ideas ? Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] phonecore, gnophone from CVS.
Hi! I was trying to use gnophone with asterisk, but I can't make a call (It just get the a answer of REJET), but I can register an everything. Anyway, I decided to move to the cvs version of gnophone, so I checked out EVERYTHING from cvs.digium.com (yes, a cvs -z7 co .). I installed libiax2, gsm (the one that was inside gnophone), and got gnophone to start compiling. But after a while, it complains about phonecore, but I can't find phonecore anywhere in the CVS. I found a couple of references about that being included in the gnophone, but it is no there (at least not in the cvs version), in the 0.2.4 there is a phonecore.c, but according to what I read, there is a phonecore package somewhere. Can anybody tell me where to find the phonecore?, or Is the gnophone under a big upgrade stage and it is not runnable from cvs rigth now? Thanks in advance for your answer, Sincerely, Ildefonso Camargo [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Identify call router? How?
I have a machine in my office, it is labelled Cable and Wireless, and on the back it says, SMarT-1 I have searched the web, and no joy. It connects to a PC via a serial cable, has anyone heard of such a device? -- * Not everyone is touched by an Angel Those that are, never forget the experience * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TDM400P??
Here is some more information about my problem: With 2 phones plugged into the 4 port FXS card, here is a situation I have witnessed: I have a clean dialtone one phone. The instant the other phone goes from on-hook to off-hook, the clean dialtone on the first line turns into a loud crackling sound with a faint dialtone in the background. I have also noticed that if I start the computer from power off with both phones plugged in, that seems to have more of a chance of normal operation (for a few minutes max), than if I plug the phones in after the machine has started up. There is also an additional problem now of the driver occasionally flooding my screen with kernel error messages. These are different error messages than the original Power alarm on module N, resetting!, (sorry I don't have the new error text available at this location). Once it starts the flooding does not stop until I unload the driver. Anyway, this problems seems to have less to do with * than it does Digium. Can anyone tell me if they have had any problems using the Digium X100P cards and the Cisco ATA186 together with asterisk?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identify call router? How?
I have a machine in my office, it is labelled Cable and Wireless, and on the back it says, SMarT-1 I have searched the web, and no joy. It connects to a PC via a serial cable, has anyone heard of such a device? Sounds like a CSU/DSU that CW installed for some service. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk using a h323 gateway
Hi all, Thank you for your help, finally we have found that it was a codec problem, now both systems are forced to use g711 ulaw and outbound calls are working fine. Best regards, Mark. -- Original Message --- From: Cerrajetto [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Fri, 12 Sep 2003 18:30:54 +0200 Subject: [Asterisk-Users] Asterisk using a h323 gateway Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 - PSTN gw)? - Asterisk ip: 192.168.1.10 - h323-PSTN gw: 192.168.1.20 I've tried: exten = _9,1,Dial(OH323/192.1.1.20) or exten = _9,1,Dial(OH323/[EMAIL PROTECTED]) but it does not work at all. If my h323 client directly uses 192.168.1.20 as h323 gateway, the calls are routed to the PSTN perfectly. What is the correct way to route some calls from Asterisk to another h323 gateway? Thank you, Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: TDM400P??
On Fri, 2003-09-19 at 10:04, Sean Rodger wrote: Can anyone tell me if they have had any problems using the Digium X100P cards and the Cisco ATA186 together with asterisk?? Yes. The only codec that is compatable with Asterisk without additional non-free codecs is the ULAW or ALAW codec. See the URL in my .sig for sample sip.conf and the various other configs, including the ATA-186 config. -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: TDM400P??
On Friday 19 September 2003 11:04 am, Sean Rodger wrote: There is also an additional problem now of the driver occasionally flooding my screen with kernel error messages. These are different error messages than the original Power alarm on module N, resetting!, (sorry I don't have the new error text available at this location). Once it starts the flooding does not stop until I unload the driver. So, do you have the P/S 4-way connector plugged into the TDM400P ? Regards...Martin -- I don't believe there really IS a GAS SHORTAGE.. I think it's all just a BIG HOAX on the part of the plastic sign salesmen -- to sell more numbers!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can Asterisk automatically initiate a call?
Hi all, Can Asterisk **initiate** a call?. If yes, what is the command? I would like that Asterisk automatically calls to me (or to somebody) and reproduces a mp3 locution, a menu, etc., is it possible? Thank you, Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Equipment listing
Hi, The following equipment is forsale on ebay: Wildcard T100P (two weeks old): http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=51279item=3048079393 Adtran TSU 600 with 12 FXO ports: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=44993item=3048077400 Cisco 7940 loaded with v5.3 SIP: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=11909item=3048080493 Cisco 7960 loaded with v5.3 SIP: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=11909item=3048080067 Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interface with PBX
Hi Folks, I'm trying to interface * with a PBX, but seems that his ring cadence is somewhat different, and my T100 doesn't show any call coming in. I've tried to change zaptel to new values but still couldn't make it work. Is there any other place where I should be changing some parameter? Is there any tool to measure the cadence timing that this pbx is providing? Thanks! PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identify call router? How?
On Fri, 2003-09-19 at 09:36, Angel Gabriel wrote: I have a machine in my office, it is labelled Cable and Wireless, and on the back it says, SMarT-1 I have searched the web, and no joy. It connects to a PC via a serial cable, has anyone heard of such a device? This isn't a flame, but maybe a quick introduction of google searching, or basically how I found the information you asked about. I went to google and searched for the terms smart-1 and router. I'll admit the results from that suck a bit, but I noticed in someones sig was a mention of the mitel smart-1 dialer. So I started a new tab and chased that rabbit hole, searching for mitel, smart-1, and dialer. This route gave me a good link. Check out this URL... http://www.mitel.co.uk/bcs/bcsprod.nsf/Pick+A+Product Under dialers you will see quite a few smart-1 devices. Look through those till you find your device. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk automatically initiate a call?
On Fri, 2003-09-19 at 10:34, Cerrajetto wrote: Hi all, Can Asterisk **initiate** a call?. If yes, what is the command? I would like that Asterisk automatically calls to me (or to somebody) and reproduces a mp3 locution, a menu, etc., is it possible? Look at sample.call in the source directory. Edit to suite and copy to /var/spool/asterisk/outgoing. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk automatically initiate a call?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Cerrajetto Sent: Friday, September 19, 2003 11:35 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can Asterisk automatically initiate a call? Hi all, Can Asterisk **initiate** a call?. If yes, what is the command? I would like that Asterisk automatically calls to me (or to somebody) and reproduces a mp3 locution, a menu, etc., is it possible? Try using Dial... From the console type 'show application Dial' for details -z ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk automatically initiate a call?
Mark- Yes, you can create a shell script that dumps a text file into /var/spool/asterisk/outgoing. Use the prototype found in /usr/src/asterisk/sample.call Name the file N.call or something similar, where N is the channel number. Create an outgoing context in your extensions.conf file to do what you want. Good luck! Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cerrajetto Sent: Friday, September 19, 2003 4:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can Asterisk automatically initiate a call? Hi all, Can Asterisk **initiate** a call?. If yes, what is the command? I would like that Asterisk automatically calls to me (or to somebody) and reproduces a mp3 locution, a menu, etc., is it possible? Thank you, Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identify call router? How?
On Fri, 2003-09-19 at 17:14, Rich Adamson wrote: I have a machine in my office, it is labelled Cable and Wireless, and on the back it says, SMarT-1 I have searched the web, and no joy. It connects to a PC via a serial cable, has anyone heard of such a device? Sounds like a CSU/DSU that CW installed for some service. How can I get it to work with my linux machines? * Not everyone is touched by an Angel Those that are, never forget the experience * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial out from script. Mini predictive dialer
Howdy, I need some pointers (ideas or help) to build a solution to retrieve recordings out of an IVR. The program needs to do some predictive dialing functions I only need it to: 1) Be able on it's own to make a call (to the same number inside this script). 2) Detect that the call has been answered and that it has finished. I need some sort of AGI but it needs to run on it's own, i.e. with no one calling the *. It will dial a number, send some DTMF's digits, record the output then detect that it has finished or hang up. Thanks, Dante ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration between *'s
Doesn't matter it should still work. Here is a hint.. dont use passwords/secrets it will then work! bkw On Fri, 19 Sep 2003, Xisco wrote: That's true if always there to connect two asterisk servers, but I'm doing some proves in order to connect one asterisk server with another SIP server. That's the matter. - Original Message - From: Jamie Carl [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 19, 2003 12:12 PM Subject: Re: [Asterisk-Users] SIP registration between *'s Why? Use IAX2, it is s much better... J On Fri, 19 Sep 2003 11:54:23 +0200 Xisco [EMAIL PROTECTED] wrote: Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =usuario1:pass1@public_ip_2 In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=public_ip_1 dtmfmode=inband Logs in * are the followings In * one logs: Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK488fe503;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as504a35d0 To: sip:usuario1@public_ip_2;tag=as2a0e47ce Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 103 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 (no NAT) topublic_ip_2:5060 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2;tag=as13445743 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 In * two logs: NOTICE[81926]: File chan_sip.c, Line 4816 (handle_request): Registration from 'sip:usuario1@public_ip_2' failed for 'public_ip_1' Sip read: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 11 headers, 0 lines Using latest request as basis request Sending to public_ip_1: 5060 (NAT) Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2;tag=as1538b8a6 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 Any idea to fix the problem Any special configuration in sip.conf Thanks a lot. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM player or plugin for XMMS
Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? Regards...Martin -- While you don't greatly need the outside world, it's still very reassuring to know that it's still there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TDM400P??
So, do you have the P/S 4-way connector plugged into the TDM400P ? Regards...Martin Yes, and I've tested the voltage to the card. Both the 12V and 5V supplies are OK to the card. -Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM player or plugin for XMMS
hi. from a shell, just type : play filename.gsm matteo. Il ven, 2003-09-19 alle 18:41, marrandy ha scritto: Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? Regards...Martin -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] exit from conference
Hi, I was trying to test the conferencing application, here is my setting in the extensions.conf exten = 5,1,MeetMe,44|p and my meetme.conf is conf = 44 but when i press the # ,itdoesn't exits my line from the conference, any suggestions Regards Azher Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software
RE: [Asterisk-Users] No sound on PSTN -- */PRI
Have you tried starting asterisk with -c? It should give you some detail as to what is happening with the call. Scott M. Stingel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Haeger Sent: Friday, September 19, 2003 3:40 PM To: Asterisk User Subject: [Asterisk-Users] No sound on PSTN -- */PRI Hi all, i tried to make a call from public pstn in our */E100P. Config is following: exten = _X.,1,Playback(testgsm) But what i hear is one dtmf tone and then nothing... Any ideas ? Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identify call router? How?
At the risk of sounding stupid. what's CSU/DSU ? *i'm googling it right now, but it's nice to have convo on the list!* A CSU/DSU is Channel Service Unit (CSU) this terminates T1 connections from the phone company. This information is then passed to the Data Service Unit which turns the signal into a serial data stream that device like a router or computer can understand. More detail definition of the terms http://www.hyperdictionary.com/dictionary/Channel+Service+Unit http://www.hyperdictionary.com/dictionary/Data+Service+Unit HTH, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM player or plugin for XMMS
Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? I just use playback in Asterisk.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM player or plugin for XMMS
You may try this one : http://www.68k.org/~michael/xmms/ It uses the audiofile library that plays many formats. On Fri, 2003-09-19 at 18:41, marrandy wrote: Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? Regards...Martin -- :: Marcel Prisi / Technical Manager --- - - - - - - - virtua.ch web solutions Ruelle du Soleil levant 6 CH - 1170 Aubonne T. +41 21 807 28 00 F. +41 21 807 28 01 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] loading dialogic drivers
Need to have chan_dialogic.so = yes in the [globals] Mark On Thu, 18 Sep 2003, pedro bulach gapski wrote: I am one of those trying to use old dialogic hardware with *. I have the following error when loading the driver: [chan_dialogic.so] = (Dialogic Global Call API Support) dlopen of libicapi.so failed: dlerror=/usr/dialogic/lib/libicapi.so: undefined symbol: gcdb_InsertLinedev WARNING[1024]: File chan_dialogic.c, Line 832 (load_module): Failed to start Global Call (GC) WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_dialogic.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_dialogic.so failed! Indeed, gcdb_InsertLinedev is not defined in libicapi. It is defined in libgc, which is linked to chan_dialogic. Anyone has seen this before? [], pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM player or plugin for XMMS
One more : http://www.zipworld.com.au/~erikd/XMMS/ This one uses libsndfile : http://www.zip.com.au/~erikd/libsndfile/ which can play even more formats including gsm6.10, G721 G723 (quite impressive) On Fri, 2003-09-19 at 18:41, marrandy wrote: Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? Regards...Martin -- :: Marcel Prisi / Technical Manager --- - - - - - - - virtua.ch web solutions Ruelle du Soleil levant 6 CH - 1170 Aubonne T. +41 21 807 28 00 F. +41 21 807 28 01 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX vs SIP
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interface with PBX
I'm trying to interface * with a PBX, but seems that his ring cadence is somewhat different, and my T100 doesn't show any call coming in. Yeah, I had a similar problem - I was trying to connect an X100P to a small 3x8 analog PBX for testing and it wouldn't grab the call. Thinking about it now, maybe I should have turned caller ID off? Hmm.. Is your T100 connected to a channel bank with FXO ports connected to PBX FXS ports? Or are you using PRI connections? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identify call router? How?
I have a machine in my office, it is labelled Cable and Wireless, and on the back it says, SMarT-1 I have searched the web, and no joy. It connects to a PC via a serial cable, has anyone heard of such a device? Sounds like a CSU/DSU that CW installed for some service. CSU = Customer Service Unit DSU = Data Service Unit Essentially, CSU/DSU's are high speed modems, converting logical bits sent from a device (eg, PC) into appropriate 56k or T1 modulation schemes that are based on telephony standards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXTel registration rejected
I have that line in my iax.conf Are you using the password they gave you when you signed up, or the new password that you were forced to pick when you logged in for the first time? I think the screen says something about it not changing your IAXTEL password, just the one you log in to the web site with, but I found this not to be the case. I log in successfully using the new password I chose. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] loading dialogic drivers
I've had problems with Dialogic apps using GlobalCall with similar symptoms, I had to type export LD_PRELOAD=/usr/dialogic/lib/libgc.so before running them. Maybe Mark's answer solves that problem also... Tim Need to have chan_dialogic.so = yes in the [globals] Mark On Thu, 18 Sep 2003, pedro bulach gapski wrote: I am one of those trying to use old dialogic hardware with *. I have the following error when loading the driver: [chan_dialogic.so] = (Dialogic Global Call API Support) dlopen of libicapi.so failed: dlerror=/usr/dialogic/lib/libicapi.so: undefined symbol: gcdb_InsertLinedev WARNING[1024]: File chan_dialogic.c, Line 832 (load_module): Failed to start Global Call (GC) WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_dialogic.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_dialogic.so failed! Indeed, gcdb_InsertLinedev is not defined in libicapi. It is defined in libgc, which is linked to chan_dialogic. Anyone has seen this before? [], pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Timothy F. Costello Sr. Systems Analyst Ne te quae siveris extra ObQuote: Sounds great! If I miss, I get to be captain. -- Chakotay to Janeway, about phasering an apple off her head, Coda ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do you get registered to IAXTEL?
Ok I know that I am new user and would like some information on how to register to use IAXTEL. I looked at the gnophone web but it does not tell me how to register or where! Yes I am very new to Asterisk and Linux so please help. I did a google on this and kept sending back to the main site IAXTEL but I feel lost! Also can someone explain what top posting is? I don't want to do this on your user list! Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How do you get registered to IAXTEL?
I looked at the gnophone web but it does not tell me how to register or where! From the main page, click on Setup (at the top), then there's a link to create a new account. Or just click here: http://gnophone.com/directory/createAccount.php Also can someone explain what top posting is? I don't want to do this on your user list! Top posting is where you hit reply, type your message, and the original message is presented below what you've typed. This is as opposed to quoted or inline replying, which I'm doing now - parts of the original message are quoted to give context to the reply which directly follows it. Top posting is considered lazy by many. I'm guilty of it in a work environment, everyone else does it (Microsoft Outlook/Exchange kinda encourages it) but in personal correspondance and mailing lists quoted replies make a lot more sense. Sure, it's a little bit more effort on your part, but it benefits everyone. Welcome to the list, welcome to Asterisk! If you want to test your IAXTEL connection drop me an email off list and we can make a couple of test calls. Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Does this thread help? http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html On Fri, Sep 19, 2003 at 01:18:53PM -0500, Peter Zeltins wrote: I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
On Fri, 19 Sep 2003, WipeOut . wrote: I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. FYI: trunking only works in IAX2 and it requires you to have a zaptel interface on both endpoints James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interface with PBX
I'm doing the following to integrate * and a Partner ACS using an 8x16 Zhone channelbank. Channel 1-4 = FXS (extensions) on Partner Chanenl 5-8 = POTS/PSTN Channel 9-16 = FXO (CO lines) on Partner This setup is working pretty well, except for a few issues with call supervision on the Zhone. Incoming calls are answered by *, then placed into a call queue that will ring into the pooled lines on the Partner system. If the caller dials an extension, * dials via one of the extensions (channel 1-4). This works well, except that it sees the line as answered immediately. If I turn on callprogress, it never sees the line answered, even when it is. For outbound, calls are routed to a 2nd * server in another location. Eventually, my inbound calls will come from the second server as well. Eventually, I'll likely drop the partner system and wire everyone directly to the Zhone (almost everyone uses cordless phones anyways). -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick Sent: Friday, September 19, 2003 2:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Interface with PBX I'm trying to interface * with a PBX, but seems that his ring cadence is somewhat different, and my T100 doesn't show any call coming in. Yeah, I had a similar problem - I was trying to connect an X100P to a small 3x8 analog PBX for testing and it wouldn't grab the call. Thinking about it now, maybe I should have turned caller ID off? Hmm.. Is your T100 connected to a channel bank with FXO ports connected to PBX FXS ports? Or are you using PRI connections? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recall doesn't seem to work
Hi I'm having a problem where the recall button doesn't work If i press recall before I dial numbers it disconnects me which is what I would expect, but during a conversation if I want to transfer the TDM 400 just ignores the recall Any advice would be gratefully received Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP + NAT Howto?
Hello Folks- Pretty new to the list here, got a lot of reading to do.. Does anyone know where I can find a decent HOWTO or set of instructions for running Asterisk and SIP clients thru firewall/NAT systems? I have a Asterisk box sitting behind a linux firewall at a remote location and have the 5060 and etc ports open as well at 16381-16391 UDP open and routed to the Asterisk box as well. I have a bunch of clients at another location which are also sitting behind a Linux ipchains/tables firewall So far, I'm able to get the clients (Xten Lite) to ring each other, but they ring, and one will say it's connected, while the other one just hangs up. -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recall doesn't seem to work
I'm having a problem where the recall button doesn't work For the benefit of our non-UK readers, recall = flashhook, but usually only 100ms timed line break, not the 500ms which seems to be the norm in North America. Robb - am I right in saying your using a UK phone? And it's definitely using timed line break (not earth recall, another european oddity)? Hmm.. This one is a bit beyond me but I'm sure there must be a timing parameter that can be set to acknowledge a shorter hook flash.. Just not sure if it's in a config file or (more likely) something that needs you to tweak the C code in one of the channel drivers. As an alternative, you can try pressing and releasing your hook switch/hang up button briefly - this may work for you. Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + NAT Howto?
I am new to * and I have been attempting to solve this same issue, but have come to the conclusion that they only way to make it work is for * to have a real reachable IP address or place another * box at the second site and use IAX trunking. This second * box, unfortunately is unsuitable for my scenario, but it may work in yours. The issue is created by the fact that the *'s real ip address is in the SDP information in the INVITE, usually this address is not directly reachable from the second site. When the XTEN actually tries to send the RTP data to this address it either dies in the network or gets an ICMP Destination Unreachable message, either way you don't have a two conversation. Side note: If you put the * on the outside the XTEN phones will have to have different RTP ports to avoid call conflicts. HTH and maybe somebody has away for this to actually work. Steve On Fri, 2003-09-19 at 16:11, C. Johnson wrote: Hello Folks- Pretty new to the list here, got a lot of reading to do.. Does anyone know where I can find a decent HOWTO or set of instructions for running Asterisk and SIP clients thru firewall/NAT systems? I have a Asterisk box sitting behind a linux firewall at a remote location and have the 5060 and etc ports open as well at 16381-16391 UDP open and routed to the Asterisk box as well. I have a bunch of clients at another location which are also sitting behind a Linux ipchains/tables firewall So far, I'm able to get the clients (Xten Lite) to ring each other, but they ring, and one will say it's connected, while the other one just hangs up. -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P question.
I am about to try our TDM400P E model from the developer kit (not the Lite) we just got and noticed a large number of reported problems. I had the CVS from Sep 12 (or so the CVS/Entries file has in it). My drivers seem to modprobe fine. My card show up as Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) in dmesg and no apparent errors...yeah. Is S400P Prototype OK for this card? My real question is physical appearance of the card. ALL the pictures show 4 modules attatched to the TDM400P (on the top edge of the card from front to back). Mine only has 1. So, Should I have 4 modules or 1? What do my missing modules do? Nothing found with the search for TDM400P via google. The current google search from the digium does not yet register any of the recent stuff yet. A search by thread or author option will be available soon! for the mailing list archives. PS My copy of the install sheets from Digium seem to ommit any reference to modprobe wcfxs when you have a TDM400P. I remembered that from previous emails I read. See, I can learn ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + NAT Howto?
Don't know yet if it helps, but if you read the link at: http://www.voip-info.org/tiki-index.php?page=NAT+and+VOIP it will point you to: http://www.sipcenter.com/files/SIPNATtraversal.pdf However has the voip-info.org site; your stuff ROCKS!! On Fri, Sep 19, 2003 at 03:11:31PM -0500, C. Johnson wrote: Hello Folks- Pretty new to the list here, got a lot of reading to do.. Does anyone know where I can find a decent HOWTO or set of instructions for running Asterisk and SIP clients thru firewall/NAT systems? I have a Asterisk box sitting behind a linux firewall at a remote location and have the 5060 and etc ports open as well at 16381-16391 UDP open and routed to the Asterisk box as well. I have a bunch of clients at another location which are also sitting behind a Linux ipchains/tables firewall So far, I'm able to get the clients (Xten Lite) to ring each other, but they ring, and one will say it's connected, while the other one just hangs up. -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] regexp problems
I'm trying to filter calls that don't have a proper ANI. This is what I did: ; only if they a real-looking ANI exten=_1XX1118/_.N.,1,Newt,1118-config ; Otherwise, send them to the loser partyline exten=_1XX1118,1,Goto(outtrunk,19096611234,1) This properly deals with null ANIs, but for some reason those with ten zeroes get matched by the first line. I also tried to be a bit more specific, like: exten=_1XX1118/_.[1-9][1-9].,1,Newt,1118-config but that also matched on all zeroes. Am I doing something wrong or is this a bug? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P question.
-= On Fri, 19 Sep 2003 15:39:44 -0500, PJ Welsh [EMAIL PROTECTED] said: I am about to try our TDM400P E model from the developer kit (not the Lite) we just got and noticed a large number of reported problems. I had the CVS from Sep 12 (or so the CVS/Entries file has in it). My drivers seem to modprobe fine. My card show up as Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) in dmesg That's odd.. mine says: Found a Wildcard FXS: Wildcard TDM400P REV E (4 modules) Try doing 'lspci'. Mine shows up as one of these two: 01:0a.0 Communication controller: Tiger Jet Network Inc. Intel 537 01:0b.0 Communication controller: Tiger Jet Network Inc. Intel 537 (I can't tell which is the FXO card.) My real question is physical appearance of the card. ALL the pictures show 4 modules attatched to the TDM400P (on the top edge of the card from front to back). Mine only has 1. So, Should I have 4 modules or 1? What do my missing modules do? The same PCI card is available with 1 to 4 FXS connections. Each connection requires one module. I think the idea is that the card could be 'expanded' later by adding modules. The RJ45 ports are already there. The Developer Kit includes one module, for one FXS port. I'm sure someone will correct me if I'm wrong, since I'm only a couple days ahead of you anyway. :) Mine has horrible amounts of static. If you plug yours in, I'd be curious to know how your audio quality turns out. PS My copy of the install sheets from Digium seem to ommit any reference to modprobe wcfxs when you have a TDM400P. Yeah, the install sheet doesn't seem to match the Developers Kit very well. I did a bit of modprobe everything until I stumbled into the correct driver. Also, the program ztcfg was new to me. That would have been something useful to put in the installation notes. You first have to cook up a zaptel.conf that looks something like this: fxsls=5 fxoks=1 loadzone=us defaultzone=us Be careful since you can NOT include ; comments like you can the other .conf files. The # is for comments in zaptel.conf only. Then run 'ztcfg' on that file: ztcfg -vvv -c /etc/asterisk/zaptel.conf That will prep the modules before you run asterisk. You probably know all that already, but maybe someone else like us will come along with a search later. -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] built in dial functions?
Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200, ata186, links to fwd and iaxtel, two x100p incoming fx lines, MoH, etc. Everything attempted to date is now working fine. However, testing the above list tends to suggest they don't work (or at least they don't work as I would expect them to.) Example, from a C7960 I dial *78# and hang up. From another sip phone I Dial that extenstion and the 7960 rings. I expected the call to roll over to voicemail or something. Am I missing something here, or are these functions not expected to work on a per-extension basis? I was assuming (probably incorrectly) these functions were custom calling features implemented within * for all extensions. Are my assumptions wrong or do I have to implement something for these to work? TIA, Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP + NAT Howto?
ClientServer XTEN -- */Firewall(NAT) ---IAX--- Firewall(NAT)/* If you are going to use IAX, I don't think you have to put * on the firewall boxes, only if you wish to use SIP. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP + NAT Howto?
Ok so if I understand correctly: For IAX, just open up the IAX ports on the firewall (the exact numbers escape me right at the moment), and let it fly? -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Varga Sent: Friday, September 19, 2003 5:27 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP + NAT Howto? Client Server XTEN -- */Firewall(NAT) ---IAX--- Firewall(NAT)/* If you are going to use IAX, I don't think you have to put * on the firewall boxes, only if you wish to use SIP. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recall doesn't seem to work
Hi. zaptel.h , line 789 #define ZT_DEFAULT_RXFLASHTIME 1250 For italy I had to lower it to 200, also be sure to lower the pulse timer (unless you're using a pulse phone with asterisk) line 792 #define ZT_MAXPULSETIME (150 * 8) I moved it to (20 * 8) be sure not to set it under ZT_MINPULSETIME, that's (15 * 8) Matteo Il ven, 2003-09-19 alle 22:04, Robert Boardman ha scritto: Hi I'm having a problem where the recall button doesn't work If i press recall before I dial numbers it disconnects me which is what I would expect, but during a conversation if I want to transfer the TDM 400 just ignores the recall Any advice would be gratefully received Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recall doesn't seem to work
I forgot...
the main problem is that eu phones seems to have flash timings
~80 - ~120 ms , so with default zaptel values, a flash hook
('R' button) is received by asterisk as one pulse, since
the pulse time is set up to 150ms ...
Matteo.
Il sab, 2003-09-20 alle 01:15, Brancaleoni Matteo ha scritto:
Hi.
zaptel.h , line 789
#define ZT_DEFAULT_RXFLASHTIME 1250
For italy I had to lower it to 200,
also be sure to lower the pulse timer
(unless you're using a pulse phone with asterisk)
line 792
#define ZT_MAXPULSETIME (150 * 8)
I moved it to (20 * 8)
be sure not to set it under ZT_MINPULSETIME, that's (15 * 8)
Matteo
Il ven, 2003-09-19 alle 22:04, Robert Boardman ha scritto:
Hi
I'm having a problem where the recall button doesn't work
If i press recall before I dial numbers it disconnects me which is what
I would expect, but during a conversation if I want to transfer the TDM
400 just ignores the recall
Any advice would be gratefully received
Thanks
Robb
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--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl
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RE: [Asterisk-Users] IAX vs SIP
How do you set up IAX in Trunk mode? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . Sent: Friday, September 19, 2003 3:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX vs SIP I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP + NAT Howto?
That's my understanding! IAX is UDP/5036 IAX2 is UDP/4569 You will probably want to use IAX2 since it can 'trunk' multiple calls in one packet. Let me know how it goes. Regards, Steve On Fri, 2003-09-19 at 18:57, C. Johnson wrote: Ok so if I understand correctly: For IAX, just open up the IAX ports on the firewall (the exact numbers escape me right at the moment), and let it fly? -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Varga Sent: Friday, September 19, 2003 5:27 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP + NAT Howto? ClientServer XTEN -- */Firewall(NAT) ---IAX--- Firewall(NAT)/* If you are going to use IAX, I don't think you have to put * on the firewall boxes, only if you wish to use SIP. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Radio for Music on Hold?
sometimes its more relevant to drop a caller into MOH with a special broadcast EG: here in cairns, we have permission during Cyclone watch etc to rebroadcast, it would be very relevant to have users listening to the local radio station whilst on hold during those times. On Fri, 19 Sep 2003 06:48:30 -0500, Peter Pauly wrote: On Thu, Sep 18, 2003 at 01:21:54PM -0700, Paul Crick wrote: Come on people! Fork out $50 for a discman and another few bucks for some royalty free library music and have that on hold instead.. You're in control, you know what your callers are listening to, and you're also legal Why go to all this trouble and expense? - skip the Discman and just rip the royaltyfree CD and save the mp3's on the hard drive. (Check the license to make sure you are allowed to do this). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] regexp problems
I'm trying to filter calls that don't have a proper ANI. This is what
I did:
; only if they a real-looking ANI
exten=_1XX1118/_.N.,1,Newt,1118-config
; Otherwise, send them to the loser partyline
exten=_1XX1118,1,Goto(outtrunk,19096611234,1)
This properly deals with null ANIs, but for some reason those with ten
zeroes get matched by the first line.
I also tried to be a bit more specific, like:
exten=_1XX1118/_.[1-9][1-9].,1,Newt,1118-config
but that also matched on all zeroes.
Am I doing something wrong or is this a bug?
Thanks...
Firstly, are you talking about caller ID's that look like 00
? Are you sure that's what they are? Try NoOp(${CALLERIDNUM}) as
your priority 1, and you should see the value in your console output.
Try exten = with the correct spacing, but that doesn't seem like it
would make a huge difference.
Have you tried with _1XX1118/_XXX. to see if you can get any
matching at all working? Be more general in what you're accepting,
and then narrow it down.
If it continues to fail, fire up a bug in the bugtracker that inludes
all the details.
JT
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[Asterisk-Users] X100 FXO Card Echo with 7960
-Original Message- From: Gary [mailto:[EMAIL PROTECTED] Sent: Friday, September 19, 2003 7:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Radio for Music on Hold? sometimes its more relevant to drop a caller into MOH with a special broadcast EG: here in cairns, we have permission during Cyclone watch etc to rebroadcast, it would be very relevant to have users listening to the local radio station whilst on hold during those times. On Fri, 19 Sep 2003 06:48:30 -0500, Peter Pauly wrote: On Thu, Sep 18, 2003 at 01:21:54PM -0700, Paul Crick wrote: Come on people! Fork out $50 for a discman and another few bucks for some royalty free library music and have that on hold instead.. You're in control, you know what your callers are listening to, and you're also legal Why go to all this trouble and expense? - skip the Discman and just rip the royaltyfree CD and save the mp3's on the hard drive. (Check the license to make sure you are allowed to do this). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I have a fair amount of echo when I call out on the FXO port of my Digium X100 using a 7960. I have played with the echo parameters in the configuration files with no change. Any suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [Release] Skinny Support in cvs
I've been poking at the Skinny channel driver since yesterday and noticed a few things. There doesn't seem to be inbound audio to the 79[46]0 phones. The status display shows a 0ms packet size. This was for calls to and from Zap, SIP and another skinny phone. One clue was that a skinny to skinny call reported the on hook event before the answered event, and the answered event was generated by app_dial. The skinny_answer does not seem to get triggered. I tried to set callerid in skinny.conf, but upon answering a call from another skinny phone, Asterisk seg-faults. (Maybe bug 264?) I found a sure-fire way to cause a segfault, and that is to dial the phone I am calling from. I know, not a real world example, but I found it by mistake and is 100% reproducable. I also noticed that the skinny.conf was not re-read by the reload command. I see bug 261 is resolved, does the status change to closed once commited to CVS, or should a CVS update from today have the fix already? I tried sending this to the dev list, but it seems to have been dropped by the moderator. Let me know if this information is not usefull. Thanks, Dan -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Saturday, September 13, 2003 9:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] [Release] Skinny Support in cvs If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s). I have personally tested this code on 7910 and 12SP+'s and will soon dive into a 7960. There currently may be issues with 7920s and ATAs, but with some proper debug information and/or the acutal device in my grubby mitts I am sure I can get around any nuances. If you have an issue with this code please use http://bugs.digium.com. Patches are absolutely apprecaited, however you should check with me before spending time as it may be a feature I have already played with locally and haven't gotten around to intergrating it into the mainline CVS code. I would like to thank miro_ for his patience and fnancial support, along with [Sim], klasstek, bkw_, PavelL, theo and ManxPower for willingly diving into nearly untested code and debuging. Lastly, we cannot forget Mark Spencer for this absolutely amazing piece of software! A quick sample config: skinny.conf: ; Typical config for a 7910 [jeremy] ; Device name device=SEP0007EB363201 ; Offical identifier (SEP+mac adress) context=default line = 500 extensions.conf: exten = 1234,1,Dial,SKINNY/[EMAIL PROTECTED]|25|r Disclaimer: All research and development of chan_skinny is for the sole purpose of writing interoperable software under Sect. 1201 (f) Reverse Engineering exception of the DMCA. The Skinny Client Control Protocol is a Cisco Systems Incorporated Trademark. chan_skinny is distributed WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IconnectHere and Ringback
When I make a call using iconnecthere, I get no ringback tone, but after the ringing the call does get connected. Any suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] built in dial functions?
These functions are implemented only for chan_zap (zaptel hardware) and work for FXS/FXO ports. Exception is *8 (remote call pickup) as far as I know. regards Martin On Fri, 19 Sep 2003, Rich Adamson wrote: Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200, ata186, links to fwd and iaxtel, two x100p incoming fx lines, MoH, etc. Everything attempted to date is now working fine. However, testing the above list tends to suggest they don't work (or at least they don't work as I would expect them to.) Example, from a C7960 I dial *78# and hang up. From another sip phone I Dial that extenstion and the 7960 rings. I expected the call to roll over to voicemail or something. Am I missing something here, or are these functions not expected to work on a per-extension basis? I was assuming (probably incorrectly) these functions were custom calling features implemented within * for all extensions. Are my assumptions wrong or do I have to implement something for these to work? TIA, Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Billing software for Asterisk?
Anyone knows there exist such a software that is working with Asterisk? Thanks = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you get registered to IAXTEL?
Try this link: http://gnophone.com/directory/createAccount.php You will find it on the setup page. The first line says: To sign up for free IAXtel access, go here. The word 'here' is the hyperlink to account creation form. As for top posting, it is what I just did here. You should also setup your email client to wrap the text at ~ 72 characters. Your email does not do this so all the text shows up on one line. HTH, Steve On Fri, 2003-09-19 at 15:03, Ariel Batista wrote: Ok I know that I am new user and would like some information on how to register to use IAXTEL. I looked at the gnophone web but it does not tell me how to register or where! Yes I am very new to Asterisk and Linux so please help. I did a google on this and kept sending back to the main site IAXTEL but I feel lost! Also can someone explain what top posting is? I don't want to do this on your user list! Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] When ISDN is busy, asterisk hangs
Hi, I have asterisk configured for german ISDN and SIP. SIP only for intranet connections. In our office there is a snom 100 and a snom 200 phone. When I'm calling a (public) telephone number which is busy, asterisk chan_modem hangs. Busy is never indicated to the calling SIP phone. And afterwords, the ISDN-channel is busy for both directions. I.e. I have to shut down asterisk and restart it in order to make any further calls or receive calls - except internal SIP calls. I'm using asterisk 0.5.0 and SuSE Linux 8.2. Any ideas? Thanks for any hints! Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTel calls coming into wrong context
I have inbound IAXtel calls working, but they come into the wrong context. I have a context= line in general above the register line in iax.conf Does anyone have any ideas what might be happening? -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budget Hotel PBX
I'm considering using asterisk to replace an existing PBX in a 40 room hotel and
would appreciate any comments, corrections or insight before I begin.
Only 8 PSTN connections are initially required but since the guests need dial-up
internet access in the rooms it has to be Frac-T1 as opposed to using FXO ports on
a channel bank.
IP phones are not an option strictly because of price. The analog phones must
have FSK message waiting lights instead of the cheaper voltage type since asterisk
doesn't support that.
So, a TE410P {or 400} and two Zhone 24FXS channel banks will be used. I
couldn't google up any info on what mobo but I'd like to start with a 450mhz since I
have one laying around with 64bit slots but if that's marginal I could get a dual
Athlon server board or whatever.
I'd also greatly appreciate knowing if anyone out there is actually using asterisk in
a
similar hotel application today.
TIA
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[Asterisk-Users] VoiceMail fromstring?
I'm having tons of trouble getting the fromstring to work in
voicemail.conf. I've tried both voicemail and voicemail2 but the emails
still seem to be coming from asterisk pbx. Has anyone had any luck with
this?
=
Here's my voicemail.conf:
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
format=wav49|gsm|wav
; Who the e-mail notification should appear to come from
serveremail=vmoperator
;[EMAIL PROTECTED]
; Should the email contain the voicemail as an attachment
attach=yes
; Maximum length of a voicemail message
;maxmessage=180
; Maximum length of greetings
;maxgreet=60
; How many miliseconds to skip forward/back when rew/ff in message
playback
skipms=3000
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence, the lower, the more
sensitive)
silencethreshold=128
; Max number of failed login attempts
maxlogins=3
; Skip the [PBX]: string from the message title
;pbxskip=yes
; Change the From: string
fromstring=VoiceMail System ; Whats wrong??? (this comment isn't here
in the real file)
; Change the email body, variables: VM_NAME, VM_DUR, VM_MSGNUM,
VM_MAILBOX, VM_CALLERID, VM_DATE
;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were
just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox
${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE} so you might\nwant to
check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
;
; Users may be located in different timezones, or may have different
; message announcements for their introductory message when they enter
; the voicemail system. Set the message and the timezone each user
; hears here. Set the user into one of these zones with the tz=
attribute
; in the options field of the mailbox. Of course, language substitution
; still applies here so you may have several directory trees that have
; alternate language choices.
;
; Look in /usr/share/zoneinfo/ for names of timezones.
; Look at the manual page for strftime for a quick tutorial on how the
; variable substitution is done on the values below.
;
; Supported values:
; 'filename'filename of a soundfile (single ticks around the
filename required)
; ${VAR}variable substitution
; A or aDay of week (Saturday, Sunday, ...)
; B or b or h Month name (January, February, ...)
; d or enumeric day of month (first, second, ..., thirty-first)
; Y Year
; I or lHour, 12 hour clock
; H Hour, 24 hour clock (single digit hours preceded by
oh)
; k Hour, 24 hour clock (single digit hours NOT preceded by
oh)
; M Minute
; P or pAM or PM
; Q today, yesterday or ABdY (*note: not standard
strftime value)
; q (for today), yesterday, weekday, or ABdY (*note:
not standard strftime value)
; R 24 hour time, including minute
;
;
[zonemessages]
eastern=America/NewYork|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H
'digits/hundred' M 'digits/hours'
;
; Each mailbox is listed in the form
mailbox=password,name,email,pager_email,options
; if the e-mail is specified, a message will be sent when a message is
; received, to the given mailbox. If pager is specified, a message will
be sent there as well.
;
[default]
100 = ,Ben Bloomberg,[EMAIL PROTECTED],tz=eastern
101 = ,Kiki Bloomberg,[EMAIL PROTECTED],tz=eastern
201 = ,Brenda Philips,[EMAIL PROTECTED],tz=eastern
202 = ,David Bloomberg,[EMAIL PROTECTED],tz=eastern
300 = ,Rosalind Philips
301 = ,Lisa Philips
302 = ,Glenda Philips
303 = ,Owen Hooks-Davis,[EMAIL PROTECTED]
Thanks
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