[Asterisk-Users] asterisk to FWD
Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf [internal] exten = 613,1,Dial(IAX2/iaxfwd-outbound/613)(service echo de FWD) exten = xx,1,Dial(IAX2/iaxfwd-outbound/xx) mon numero FWD exten = yy,1,Dial(IAX2/iaxfwd-outbound/yy) celui d'un ami FWD iax.conf [general] context=default bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no tos=lowdelay autokill=yes allow=ulaw language=fr register = xx:[EMAIL PROTECTED] [iaxfwd-outbound] type=peer username=xx host=fwd.pulver.com secret=mon_passwd_FWD disallow=all allow=ulaw allow=gsm allow=ilbc allow=g726 nat=yes when I call the 613 number (echo FWD service), I have this message from my PBX: Executing Dial(SIP/xlite-9f55, IAX2/iaxfwd-outbound/613) in new stack -- Called iaxfwd-outbound/613 Feb 7 09:38:17 NOTICE[2744]: chan_iax2.c:2821 auto_congest: Auto-congesting call due to slow response -- IAX2/iaxfwd-outbound-1 is circuit-busy -- Hungup 'IAX2/iaxfwd-outbound-1' == Everyone is busy/congested at this time (1:0/1/0) Please, how can I resolve this problem? Thank you very much -- Bayrouni ___ --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] asterisk to FWD
One problem I can see is that you're not using the keys that come with
asterisk.
Mine (which works!) looks like this
iax.conf
register = user:[EMAIL PROTECTED]
[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
disallow=all
allow=ulaw
qualify=yes
extensions.conf
; Calls to FWD
exten = _393.,1,Set(CALLERID=37720)
exten = _393.,2,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:3}|20)
exten = _393.,3,Congestion
[from-fwd]
exten = 37720,1,SetCallerID(393${CALLERIDNUM})
exten = 37720,2,Dial(SIP/2208,20)
exten = 37720,3,Voicemail,u2208
exten = 37720,4,Hangup
exten = 37720,103,Voicemail,b2208
exten = 37720,104,Hangup
Try this and see how it goes.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Bayrouni wrote:
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
[internal]
exten = 613,1,Dial(IAX2/iaxfwd-outbound/613)(service echo de FWD)
exten = xx,1,Dial(IAX2/iaxfwd-outbound/xx) mon numero FWD
exten = yy,1,Dial(IAX2/iaxfwd-outbound/yy) celui d'un ami FWD
iax.conf
[general]
context=default
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
allow=ulaw
language=fr
register = xx:[EMAIL PROTECTED]
[iaxfwd-outbound]
type=peer
username=xx
host=fwd.pulver.com
secret=mon_passwd_FWD
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
allow=g726
nat=yes
when I call the 613 number (echo FWD service), I have this
message from my PBX:
Executing Dial(SIP/xlite-9f55, IAX2/iaxfwd-outbound/613) in new stack
-- Called iaxfwd-outbound/613
Feb 7 09:38:17 NOTICE[2744]: chan_iax2.c:2821 auto_congest:
Auto-congesting call due to slow response
-- IAX2/iaxfwd-outbound-1 is circuit-busy
-- Hungup 'IAX2/iaxfwd-outbound-1'
== Everyone is busy/congested at this time (1:0/1/0)
Please, how can I resolve this problem?
Thank you very much
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Re: [Asterisk-Users] asterisk to FWD
I forgot to add that you must have an IAX acount with FWD. A regular SIP
account won't let you then use IAX. You have to register for it.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Mark Phillips wrote:
One problem I can see is that you're not using the keys that come with
asterisk.
Mine (which works!) looks like this
iax.conf
register = user:[EMAIL PROTECTED]
[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
disallow=all
allow=ulaw
qualify=yes
extensions.conf
; Calls to FWD
exten = _393.,1,Set(CALLERID=37720)
exten = _393.,2,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:3}|20)
exten = _393.,3,Congestion
[from-fwd]
exten = 37720,1,SetCallerID(393${CALLERIDNUM})
exten = 37720,2,Dial(SIP/2208,20)
exten = 37720,3,Voicemail,u2208
exten = 37720,4,Hangup
exten = 37720,103,Voicemail,b2208
exten = 37720,104,Hangup
Try this and see how it goes.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Bayrouni wrote:
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
[internal]
exten = 613,1,Dial(IAX2/iaxfwd-outbound/613)(service echo de FWD)
exten = xx,1,Dial(IAX2/iaxfwd-outbound/xx) mon numero FWD
exten = yy,1,Dial(IAX2/iaxfwd-outbound/yy) celui d'un ami FWD
iax.conf
[general]
context=default
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
allow=ulaw
language=fr
register = xx:[EMAIL PROTECTED]
[iaxfwd-outbound]
type=peer
username=xx
host=fwd.pulver.com
secret=mon_passwd_FWD
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
allow=g726
nat=yes
when I call the 613 number (echo FWD service), I have this
message from my PBX:
Executing Dial(SIP/xlite-9f55, IAX2/iaxfwd-outbound/613) in new
stack
-- Called iaxfwd-outbound/613
Feb 7 09:38:17 NOTICE[2744]: chan_iax2.c:2821 auto_congest:
Auto-congesting call due to slow response
-- IAX2/iaxfwd-outbound-1 is circuit-busy
-- Hungup 'IAX2/iaxfwd-outbound-1'
== Everyone is busy/congested at this time (1:0/1/0)
Please, how can I resolve this problem?
Thank you very much
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Re: [Asterisk-Users] asterisk to FWD
Mark Phillips wrote:
One problem I can see is that you're not using the keys that come with
asterisk.
Mine (which works!) looks like this
iax.conf
register = user:[EMAIL PROTECTED]
[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
disallow=all
allow=ulaw
qualify=yes
extensions.conf
; Calls to FWD
exten = _393.,1,Set(CALLERID=37720)
exten = _393.,2,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:3}|20)
exten = _393.,3,Congestion
[from-fwd]
exten = 37720,1,SetCallerID(393${CALLERIDNUM})
exten = 37720,2,Dial(SIP/2208,20)
exten = 37720,3,Voicemail,u2208
exten = 37720,4,Hangup
exten = 37720,103,Voicemail,b2208
exten = 37720,104,Hangup
Try this and see how it goes.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Bayrouni wrote:
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
[internal]
exten = 613,1,Dial(IAX2/iaxfwd-outbound/613)(service echo de FWD)
exten = xx,1,Dial(IAX2/iaxfwd-outbound/xx) mon numero FWD
exten = yy,1,Dial(IAX2/iaxfwd-outbound/yy) celui d'un ami FWD
iax.conf
[general]
context=default
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
allow=ulaw
language=fr
register = xx:[EMAIL PROTECTED]
[iaxfwd-outbound]
type=peer
username=xx
host=fwd.pulver.com
secret=mon_passwd_FWD
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
allow=g726
nat=yes
when I call the 613 number (echo FWD service), I have this
message from my PBX:
Executing Dial(SIP/xlite-9f55, IAX2/iaxfwd-outbound/613) in new
stack
-- Called iaxfwd-outbound/613
Feb 7 09:38:17 NOTICE[2744]: chan_iax2.c:2821 auto_congest:
Auto-congesting call due to slow response
-- IAX2/iaxfwd-outbound-1 is circuit-busy
-- Hungup 'IAX2/iaxfwd-outbound-1'
== Everyone is busy/congested at this time (1:0/1/0)
Please, how can I resolve this problem?
Thank you very much
___
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Thank you,
yes,
there was problems with some keys.
the secret was incorrect and host was too incorrect.
Thanks
a +
--
Bayrouni
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Re: [Asterisk-Users] Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP
cmisip wrote: No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1) I don't know why the above message is printing codec numnbers, rather than names. *shrug* show codecs will tell you what codec number are what codec name. It appears that your Phone/phone0 is using G723.1. Looks likes one of the newbie problems of using allow=all or bandwidth=low. DON'T DO THAT! Use disallow=all and then allow= lines for the one or more codecs that you actually want to use. Asterisk does not fully support G723.1. fully means transcode. --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP
I cant seem to be able to figure this out. As much as I can tell it is a codec problem. I can dial out to [EMAIL PROTECTED] and the Call Me test there rings my phone. However when the callee endpoint answers, there is a failure to translate: Outgoing Call for 612 612 is not a local user -- Called [EMAIL PROTECTED] No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1) (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/fwdpulvercom-dd5a is ringing Unable to handle indication 3 for 'Phone/phone0' Scheduled a registration timeout # 100 Acked pending invite 102 Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found build_route: Record-Route hop: sip:[EMAIL PROTECTED];ftag=as3d6e380d;lr=on build_route: Contact hop: sip:[EMAIL PROTECTED]:5028 -- SIP/fwdpulvercom-dd5a answered Phone/phone0 No path to translate from Phone/phone0(1) to SIP/fwdpulvercom-dd5a(2) Had to drop call because I couldn't make Phone/phone0 compatible with SIP/fwdpulvercom-dd5a update_user_counter(612) - decrement outUse counter I have a Quicknet Lite ISA card. my phone.conf contains: mode=dialtone ;format=slinear format=g723.1 echocancel=medium silencesupression=yes device = /dev/phone0 my sip.conf contains: context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register = 6:[EMAIL PROTECTED] [fwdout] type=friend username=6 secret=mypasswd host=fwd.pulver.com [fwdin] type=peer host=fwd.pulver.com context=default nat=yes canreinvite=no my extensions.conf contains: [globals] CONSOLE=Phone/phone0 [default] exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = s,1,Dial(Phone/phone0) Is it possible to call FWD using the Quicknet card? Thanks for any help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, newbie, fwd and is this jitter?
Hi I've been playing around with asterisk for a while now at home, just trying to understand a bit of the technology and seeing what I could get up and running. Here's where I am at: I bought myself an X100P card and got an asterisk server up and running on a gentoo linux distro. I got two windows PC clients, running x-lite to work with this and got basic functionality running. After some help from this mailing list I was able to add festival and music on hold to asterisk and this all seems ok. I then looked at connecting asterisk to FWD. First I set it all up as a regular SIP connection and then I moved it onto using IAX. However no matter how I connect to FWD, festival and music on hold sound awful but fine over the analogue connection. If anyone has FWD running, is what you hear on 290718 as good as it gets? Is this jitter that appears in posts to the mailing list? Regards Quentin BTW, I've disconnected the analogue just in case my dial plan is not too clever ;-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and fwd
Shoval Tomer wrote : Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com and it doesn't work. Not even when calling to 5. I presume you're looking at the asterisk console (ie. started asterisk with option -vvvc) ? Can you advise on how to debug sip (or trace and view sip packets) from the asterisk server to fwd so we can try to understand what exactly is not working? If the regisstration went correctly you should see 'normal' call behaviour in your client, the client calls the number and thinks the call is connected. Also you can issue : 'sip show peers'. It should give you something like : *CLI sip show peers Name/usernameHost Mask Port Status sipphone/xx 130.94.123.252 255.255.255.255 5060 Unmonitored xlite(Unspecified) (D) 255.255.255.255 0UNKNOWN fwd/username 192.246.69.223 255.255.255.255 5060 OK (155 ms) Please be advised that our asterisk server is behind a NAT firewall. I'm guessing you have succesfully opened the portranges needed in your fw config? If so you also need the sip-nat patch by Leif Madsen. There has been a topic for the last two weeks (subject : [Asterisk-Users] Asterisk behind NAT How to do it.). Look at this thread, install the patch and you should be fine. Arnold. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and fwd
Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com and it doesn't work. Not even when calling to 5. Can you advise on how to debug sip (or trace and view sip packets) from the asterisk server to fwd so we can try to understand what exactly is not working? Please be advised that our asterisk server is behind a NAT firewall. Thanks in advance, Tom.
Re: [Asterisk-Users] Asterisk and fwd
If U want I can send U my settings. My FWD is working fine. Let me know... Chris HARIGA - Original Message - From: Shoval Tomer To: [EMAIL PROTECTED] Sent: Sunday, December 14, 2003 10:31 AM Subject: [Asterisk-Users] Asterisk and fwd Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com and it doesn't work. Not even when calling to 5. Can you advise on how to debug sip (or trace and view sip packets) from the asterisk server to fwd so we can try to understand what exactly is not working? Please be advised that our asterisk server is behind a NAT firewall. Thanks in advance, Tom.
Re: [Asterisk-Users] Asterisk and fwd
Shoval Tomer wrote: Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com http://www.loligo.com/ and it doesn't work. Not even when calling to 5. Check the Asterisk FAQ at http://www.voip-info.org Can you advise on how to debug sip (or trace and view sip packets) from the asterisk server to fwd so we can try to understand what exactly is not working? Test the CLI command sip debug. Regards, /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and fwd
Please do. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris HARIGA Sent: Sunday, December 14, 2003 8:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk and fwd If U want I can send U my settings. My FWD is working fine. Let me know... Chris HARIGA - Original Message - From: Shoval Tomer To: [EMAIL PROTECTED] Sent: Sunday, December 14, 2003 10:31 AM Subject: [Asterisk-Users] Asterisk and fwd Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com and it doesn't work. Not even when calling to 5. Can you advise on how to debug sip (or trace and view sip packets) from the asterisk server to fwd so we can try to understand what exactly is not working? Please be advised that our asterisk server is behind a NAT firewall. Thanks in advance, Tom.
RE: [Asterisk-Users] Asterisk and FWD
Did you move your box behind some type of nat device?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
Sent: Tuesday, July 22, 2003 10:06 AM
To: Asterisk Users
Subject: [Asterisk-Users] Asterisk and FWD
Hi,
The line
register = fwdnr:[EMAIL PROTECTED]/101
does not work anymore in sip.con file.
I get the following error in the Asterisk console:
-- Got SIP response 479 We dont accept private IP contacts. please Set
your external IP back from 192.246.69.223
The sane error when I try to call a FWD extension defined like that:
Something changed in the mean time?
exten = _X,1,SetCallerID(${FWDUSERID})
exten = _X,2,SetCIDName(${FWDUSERNAME})
exten = _X,3,Dial(SIP/[EMAIL PROTECTED])
exten = _X,4,Hangup
exten = _X,104,Hangup
Thanks,
Dan
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[Asterisk-Users] Asterisk and FWD
I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register = [EMAIL PROTECTED]/1 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=1 fromuser=1 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to mailbox: exten = 71,1,Dial(SIP/[EMAIL PROTECTED]) are bounced back with 403 Forbidden What I'm doing wrong there? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
