Re: [asterisk-users] Help with MusicOnHold
Do you have the mpg123 utility in your system? By default, Asterisk uses mpg123 to play the mp3 files for music on hold. -kokmeng. Julian Varanini wrote: Hi, I am running asterisk 1.1. When a client is placed on hold from the x-lite or polycom phone, no hold music is heard. I have musicclass=default set up in sip.conf and default exists in musiconhold.conf. Has anyone had a similar experience? Any help would be appreciated. Thanks Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel.conf settings for Singtel ISDN-2
Hi, Has anyone successfully configured a HFC ISDN card with Singtel's ISDN-2 service? If so, can you share the settings required? Thanks in advance. KokMeng. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel.conf settings for Singtel ISDN-2
Hi Leo, How stupid of me! I just realized that I needed a NT-1 box between my HFC card and the ISDN line! What I observed was that the line was always not active. Thanks for your reply anyway. -kokmeng. Leo Ann Boon wrote: KokMeng Loh wrote: Hi, Has anyone successfully configured a HFC ISDN card with Singtel's ISDN-2 service? If so, can you share the settings required? The AVM Fritz! PCI works fine with chan_capi for Singtel. At that time (about 2 years back), I didn't test HFC because the driver was very immature. What kind of problems do you have? Did you try to connect through a TA box (the NT-1) or direct? Cheers. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN in Japan
Hi, Has anyone gotten the Billion BiPAC PCI ISDN card to work with the ISDN (INS net64) and Asterisk in Japan? http://www.billion.com/product/isdn/bipacpciv30.htm Regards, KokMeng. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA3000 in Singapore
Hi, I need help getting my SPA3000 to work with the line settings in Singapore. Has anyone gotten it to work in Singapore? Regards, KokMeng Loh ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Major Problems UTStarcom F1000 registering -- pls help
Hi, You can try changing your section name ([UTStarcomF1000]) to the user name, i.e. [anonymous]. I also noticed that you had a typo in the 'dtmfmode' line; it should be 'rfc2833' and not 'rfca2833'. -kokmeng. Christoph Merk wrote: Hi there, I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with my asterisk server. I already changed the name of the user to anonymous since it looks like the phone sends that name. The WiFi Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200 What is it that I am missing? Any help very much appreciated!!! The error message I get is: Jan 11 13:49:30 NOTICE[24024]: chan_sip.c:10817 handle_request_register: Registration from 'anonymous sip:[EMAIL PROTECTED]' failed for '192.168.1.217' - Username/auth name mismatch extract of [sip.conf]: ... [UTStarcomF1000]type=friend bindport=5060 username=anonymous ;fromuser=anonymous secret=welcome mailbox=1000 canreinvite=yes context=sipout insecure=very defaultip=192.168.1.217 host=dynamic qualify=yes nat=no ;auth=anonymous:[EMAIL PROTECTED] dtmfmode=rcfa2833 *CLI sip show peers Name/username HostDyn Nat ACL Port Status UTStarcomF1000/anonymous (Unspecified)D 0UNKNOWN omp-out-4321/419941x 212.117.200.148 N 5060 OK (64 ms) omp-out-5211/419941x 212.117.200.148 N 5060 OK (64 ms) omp-out-5200/419941x 212.117.200.148 N 5060 OK (64 ms) web-de/x 217.72.200.89N 5060 OK (64 ms) sipgate-out/19x217.10.79.9 N 5060 OK (68 ms) 8 sip peers [5 online , 3 offline] *CLI sip debug ip 192.168.1.217 SIP Debugging Enabled for IP: 192.168.1.217 *CLI sip show registry HostUsername Refresh State sip.web.de:5060 x 105 Registered sipgate.de:5060 19x105 Registered And here the debug message: . Jan 11 14:28:38 NOTICE[24049]: chan_sip.c:10817 handle_request_register: Registration from 'anonymous sip:[EMAIL PROTECTED] ' failed for '192.168.1.217' - Username/auth name mismatch Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from 192.168.1.217:5060: REGISTER sip:192.168.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.217:5060;rport;branch=z9hG4bK3499846672 From: anonymous sip:[EMAIL PROTECTED];tag=787472657 To: anonymous sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 90 REGISTER Contact: sip:[EMAIL PROTECTED]:5060;action=proxy max-forwards: 70 expires: 60 user-agent: UTSTARCOM F1000/Device ID-0007ba253307 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.1.217 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.1.217:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.1.217:5060;rport;branch=z9hG4bK3499846672;received=192.168.1.217 From: anonymous sip:[EMAIL PROTECTED];tag=787472657 To: anonymous sip:[EMAIL PROTECTED];tag=as750293ee Call-ID: [EMAIL PROTECTED] CSeq: 90 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Content-Length: 0 and here is the SIP and RTP Configuration of the phone: (STUN is turned off) (I hope this will be transmitted to the list as well since it is a paste from the Web Interfrace. In short it says: Sip Terminal Use Outbound Proxy yes sip terminal use register yes sip outbound server domain name server.x.y sip outbound server ip address 192.168.1.200 sip outbound server port 5060 sip rigister server domain name server.x.y sip register server ip address 192.168.1.200 sip register server port 5060 sip authentication string anonymous sip user name anonymous sip password welcome sip terminal port 5060 sip terminal use null packet no both sip proxy and regisister server use IP yes dns query type yes set registration duration 60 sec terminal audio rtp port 10120 terminal audio packetize time 20 milliseconds *SIP Terminal Use Outbound Proxy:* No Yes *SIP Terminal Use Register: * No Yes *SIP Outbound Server Domain Name:* *SIP Outbound Server IP Address:* *SIP Outbound Server Port:* *SIP Register Server Domain Name:* *SIP Register Server IP Address:* *SIP Register Server Port:* *SIP Authentication String:* *SIP User Name:* *SIP Password:* *SIP Terminal Port:* *SIP Terminal Use Null Packet:* No Yes *SIP Terminal Use DNS:* Both SIP Proxy And Register Servers Use IP Register Server Uses DNS And SIP Proxy Uses IP Register Server Uses IP And SIP Proxy Server Uses DNS Both Register And SIP Proxy Servers Use
Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu
Hi Aisling,
You're missing the 'WaitExten' directive after playing the background
sound file. Your lines should be like this:
[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = s,n,WaitExten(10)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)
-kokmeng.
Aisling wrote:
Hi,
Thanks to both Iqbal and Kokmeng for the replies.
Kokmeng I tried what you suggested however no luck...
What I have done which is currently working(kind of) is that in my
sip.conf in the [general] section I have set context=incomingpstn. My
register line looks like:
register = username:[EMAIL PROTECTED]/
In my extensions.conf I then have
[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)
[internalExt]
exten = s,n,Background(InternalExtension)
[mainconfmenu]
exten = s,n,Background(MainConfMenu)
I can hear the MainMenu sound file being played. What's strange is that
when I press '1' to interrupt, which in my logic should invoke the
internalExt context, nothing happens. The MainMenu sound file continues
to play and finally I get the error:
Warning: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
'incomingpstn'
I used the 'Goto' as Iqbal suggested instead of includes...
Has anyone ever experienced this kind of behaviour before?
Many thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 09 January 2006 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Stumped
Hi,
The hostname that you used in your register directive ('provider.ie')
must have a corresponding section in sip.conf. In your example, you used
'[provider-in]'. If that is what you actually used, then this might
explain why your incoming goes to the default context because it
couldn't find its own section. Try renaming '[provider-in]' to
'[provider.ie]'.
-kokmeng.
Aisling O'Driscoll wrote:
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd that's happening (and I'm very stumped
with this)..I think my contexts are definately the reason that I
can't interrupt the menu for incoming pstn calls to choose a submenu:
My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a 'dummy' extension.
sip.conf
register = username:[EMAIL PROTECTED]/2093
[provider-in]
type=peer
host=sip.provider.ie
context=onecontext
[2092]
type=peer
other stuff
context=onecontext
So the dummy extension here is '2093' and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
'onecontext' context.
Now in my extensions.conf 'onecontext' includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of 'onecontext' is to allow outgoing access to the
PSTN.
[onecontext]
include = createmenu//creating an IVR menu
include = createconf//creating a conf call
etc
include = default //used for voicemail
[createmenu]
;does something
[createconf]
;does something
;outgoing calls - main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup
[default]
;mailbox for 2092 and other users
Now this is where the problems start! For incoming calls I tried to
do include = incomingpstn in 'onecontext' which I thought would
call a new context called 'incomingpstn' which would have an entry
for the dummy user. i.e.
[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension
I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn't work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the 'incomingpstn' context
to the default context i.e.
[default]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension
With this I can hear the MainMenu when I dial my DDI but I can't seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don't have access to the other features in Asterisk.
The point is that I'm stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.
This is what comes up on my asterisk console:
-- Executing Dial
Re: [Asterisk-Users] Incoming PSTN Calls - Stumped
Hi,
The hostname that you used in your register directive ('provider.ie')
must have a corresponding section in sip.conf. In your example, you used
'[provider-in]'. If that is what you actually used, then this might
explain why your incoming goes to the default context because it
couldn't find its own section. Try renaming '[provider-in]' to
'[provider.ie]'.
-kokmeng.
Aisling O'Driscoll wrote:
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd that’s happening (and I’m very stumped
with this)….I think my contexts are definately the reason that I
can’t interrupt the menu for incoming pstn calls to choose a submenu:
My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a ‘dummy’ extension.
sip.conf
register = username:[EMAIL PROTECTED]/2093
[provider-in]
type=peer
host=sip.provider.ie
context=onecontext
[2092]
type=peer
other stuff
context=onecontext
So the dummy extension here is ‘2093’ and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
‘onecontext’ context.
Now in my extensions.conf ‘onecontext’ includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of ‘onecontext’ is to allow outgoing access to the
PSTN.
[onecontext]
include = createmenu//creating an IVR menu
include = createconf//creating a conf call
etc
include = default //used for voicemail
[createmenu]
;does something
[createconf]
;does something
;outgoing calls – main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup
[default]
;mailbox for 2092 and other users
Now this is where the problems start! For incoming calls I tried to
do “include = incomingpstn” in ‘onecontext’ which I thought would
call a new context called ‘incomingpstn’ which would have an entry
for the dummy user. i.e.
[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension
I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn’t work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the ‘incomingpstn’ context
to the default context i.e.
[default]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension
With this I can hear the MainMenu when I dial my DDI but I can’t seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don’t have access to the other features in Asterisk.
The point is that I’m stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.
This is what comes up on my asterisk console:
-- Executing Dial (“SIP/2092-2829”, “SIP/[EMAIL PROTECTED]) in
new stack
-- Called [EMAIL PROTECTED]
-- Playing ‘MainMenu’ (language ‘en’)
-- other messages (not relevant I think)
== Spawn extension (outgoing, 021123456, 1) exited non-zero on
‘SIP/2092-5837’
== Spawn extension (default, 2093, 2) exited non zero etc etc
I’m very stuck on this and can’t figure it out.
Any help appreciated.
Many thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giovanni Miano
Sent: 05 January 2006 21:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls
Is Exist InternalExtension context ? and 2093 exten ?
2006/1/5, Aisling [EMAIL PROTECTED]:
Hi all,
I am having difficulty getting incoming PSTN calls working. I have
set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
My provider told me to change my sip.conf as follows
register = username:[EMAIL PROTECTED]/2093
; To receive incoming calls specify this block and replace
yourcontext for your dial plan.
[blueface-in]
type=peer
host=sip.blueface.ie
context=incomingpstn
And then in my extensions.conf to have something similar to the
following (or however I wanted to handle my incoming calls)
[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)
//press 1 for internal extensions.
This didn't work and I kept getting a 404 not found error saying the
user didn't exist. I tried creating the user in sip.conf and
