I keep getting the following Auto-congesting message whenever I try to dial from
an X-Lite SIP phone to another one within my LAN. It's a real basic
configuration but I am unable to figure out what is happening:
localhost*CLI>
-- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in
new stack
-- Called jtest
May 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019 auto_congest: Auto-congestin
g SIP/jtest-6a1e
-- SIP/jtest-6a1e is circuit-busy
== Everyone is busy at this time
May 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No applica
tion 'DialCongestion' for extension (sip, 22, 2)
== Spawn extension (sip, 22, 2) exited non-zero on 'SIP/jay-de1b'
My setup is very simple and basic:
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = sip; Default
[jay]
type=friend
secret=jaysip
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=100
dtmfmode=inband
callerid="Jay <400>"
disallow=all
allow=gsm
context=sip
[jtest]
type=friend
secret=jaytestsip
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=100
dtmfmode=inband
callerid="Jay Test <410
disallow=all
allow=gsm
context=sip
extensions.conf
[sip] ; context for X-Lite Clients
exten =>11,1,Dial(SIP/jay,20,tr)
exten =>11,2,Congestion
exten =>22,1,Dial(SIP/jtest,20,tr)
exten =>22,2,DialCongestion
Lastly, here's my client setup
Display Name: Jay
User Name & Authorization User: jay
Password: jaysip
Domain/Realm: 192.168.1.20
SIP Proxy: 192.168.1.20
Display Name: Jay Test
User Name & Authorization User: jtest
Password: jaytestsip
Domain/Realm: 192.168.1.20
SIP Proxy: 192.168.1.20
Any help anyone can give me would be appreciated since I've already spent
HOURS on this and have made absolutely no progress in debugging this (didn't
find anything in any of the archives nor wiki pages).
J...
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Jeremy McNamara
2004-May-02 09:11 UTC
[Asterisk-Users] Simple SIP X-Lite Configuration Failing
J Poz wrote:> -- SIP/jtest-6a1e is circuit-busy > == Everyone is busy at this time> May 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: > No applica > tion 'DialCongestion' for extension (sip, 22, 2)Come on... Asterisk is telling you EXACTLY whats wrong... SIP/jtest is circuit-busy and then asterisk cannot find an application named DialCongetsion. Fix your extensions.conf Jeremy McNamara
Can anyone help. I've changed the extensions.conf file as follows:
extensions.conf
[sip] ; context for X-Lite Clients
exten =>11,1,Dial(SIP/jay,20,tr)
exten =>22,1,Dial(SIP/jtest,20,tr)
I'm still getting the Auto-congesting error (and circuit-busy). Does anyone
know what is causing this in such a simple configuration?
localhost*CLI>
-- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in
new stack
-- Called 410
May 2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest:
Auto-congesting SIP/410-a4a1
-- SIP/410-a4a1 is circuit-busy
== Everyone is busy at this time
J Poz <jpoz0000@yahoo.com> wrote:
I keep getting the following Auto-congesting message whenever I try to dial from
an X-Lite SIP phone to another one within my LAN. It's a real basic
configuration but I am unable to figure out what is happening:
localhost*CLI>
-- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in
new stack
-- Called jtest
May 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019 auto_congest: Auto-congestin
g SIP/jtest-6a1e
-- SIP/jtest-6a1e is circuit-busy
== Everyone is busy at this time
May 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No applica
tion 'DialCongestion' for extension (sip, 22, 2)
== Spawn extension (sip, 22, 2) exited non-zero on 'SIP/jay-de1b'
My setup is very simple and basic:
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = sip; Default
[jay]
type=friend
secret=jaysip
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=100
dtmfmode=inband
callerid="Jay <400>"
disallow=all
allow=gsm
context=sip
[jtest]
type=friend
secret=jaytestsip
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=100
dtmfmode=inband
callerid="Jay Test <410
disallow=all
allow=gsm
context=sip
extensions.conf
[sip] ; context for X-Lite Clients
exten =>11,1,Dial(SIP/jay,20,tr)
exten =>11,2,Congestion
exten =>22,1,Dial(SIP/jtest,20,tr)
exten =>22,2,DialCongestion
Lastly, here's my client setup
Display Name: Jay
User Name & Authorization User: jay
Password: jaysip
Domain/Realm: 192.168.1.20
SIP Proxy: 192.168.1.20
Display Name: Jay Test
User Name & Authorization User: jtest
Password: jaytestsip
Domain/Realm: 192.168.1.20
SIP Proxy: 192.168.1.20
Any help anyone can give me would be appreciated since I've already spent
HOURS on this and have made absolutely no progress in debugging this (didn't
find anything in any of the archives nor wiki pages).
J...
---------------------------------
Do you Yahoo!?
Win a $20,000 Career Makeover at Yahoo! HotJobs
---------------------------------
Do you Yahoo!?
Win a $20,000 Career Makeover at Yahoo! HotJobs
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Girish Gopinath
2004-May-05 10:42 UTC
[Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing
Hello, Replying to the mail which was posted 3 days back. I tested the configuration here with SJphones, and got the same error: "circuit-busy". I tried with sip debug turned on, and found that asterisk receives a CANCEL request from the user agent immediately after it receives INVITE. When i first saw this mail, i thought it was a simple config issue, but even after trying for more than 2 hours, i am not able to figure out why it is happening. I tried changing the sip.conf entries with the minimum required values, but no success. I started evaluating Asterisk a few months ago, i also tried with such simple configurations and did not have issues like this. Here is my Asterisk version: Asterisk CVS-02/21/04-16:21:31 built by root@localhost.localdomain on a i686 running Linux I am really curious if you were able to solve the problem. If so, what was the reason behind that weird behaviour and how did you solve it? If not, can anyone please tell what is going wrong? Regards, Girish BTW, J Poz, dont use reinvite=, it does not exist, use canreinvite= instead.>From: J Poz <jpoz0000@yahoo.com> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing >Date: Sun, 2 May 2004 16:41:52 -0700 (PDT) > >Sorry for any confusion.........But in my latest error, instead of calling >my clients "jay" and "jtest", I'm calling them "400" and "410".. Everything >else is still the same and it's same problem. > >My guess is that I've set a parameter incorrectly and therefore Asterisk >thinks there's only one client so any calls I try to make between the two >fail since it thinks the other client is busy. But I don't understand >enough to interpret the error message. I thought the SIP part would be the >easy part - I already have the FXO and FXS interfaces working. > >Again, thanks for anyone who can help me since I am at a loss! > >J Poz <jpoz0000@yahoo.com> wrote: >Can anyone help. I've changed the extensions.conf file as follows: > >extensions.conf >[sip] ; context for X-Lite Clients >exten =>11,1,Dial(SIP/jay,20,tr) >exten =>22,1,Dial(SIP/jtest,20,tr) > >I'm still getting the Auto-congesting error (and circuit-busy). Does anyone >know what is causing this in such a simple configuration? > > >localhost*CLI> > -- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in new stack > -- Called 410 >May 2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest: >Auto-congesting SIP/410-a4a1 > -- SIP/410-a4a1 is circuit-busy > == Everyone is busy at this time >_________________________________________________________________ Send flowers in 24 hours! http://www.fabmall.com/affiliatehtml/redir/nl7.asp At MSN Shopping.
Girish and All, I got my SIMPLE SIMPLE X-Lite to X-Lite configuration to work. First, I want to thank "William Ray" who helped me offline. I found him by one of his postings in the http://asterisk.xvoip.com/ forum where he helped someone else with a similar issue. He gave me a working configuration for what I was trying to do and it worked. However, I still needed to debug and find out what was wrong with my configuration since I know it can help someone else in the future (if they search the mailing list for the same or similar problem). The main problem that PREVENTED the X-Lite softphones from communicating with eachother was I had invalid syntax in the sip.conf file for the caller-id field as follows: Incorrect Syntax: callerid="Jay <400>" Correct Syntax: callerid="Jay" <400> It blows my mind why asterisk would spit out "Auto-congesting" and "is circuit-busy" errors since I don't know how they're related. That was the main problem that needed to get fixed for the configuration to work. Also, I then changed the DTMF mode to "rfc2833" from "inband" to get rid of "Unable to process inband DTMF on 2 frames" warnings. Everything is PEACHY now.. I hope this proves useful to others in the future. I invested over 30 hours on this problem so hopefully it can be avoided by someone else. J.......... J Poz <jpoz0000@yahoo.com> wrote: Girish, Thanks for replying and trying to work my "simple configuration". Nobody on the list has replied with any help and I still have the problem. I've invested well over 20 hours on this problem and still don't have a solution (I have everything else within Asterisk working including IVR menus, X100 interfaces, etc). However, I am not able to get a simple Softphone to Softphone configuration to work. Can anyone on the LIST help us Girish Gopinath <gopinath_girish@hotmail.com> wrote: Hello, Replying to the mail which was posted 3 days back. I tested the configuration here with SJphones, and got the same error: "circuit-busy". I tried with sip debug turned on, and found that asterisk receives a CANCEL request from the user agent immediately after it receives INVITE. When i first saw this mail, i thought it was a simple config issue, but even after trying for more than 2 hours, i am not able to figure out why it is happening. I tried changing the sip.conf entries with the minimum required values, but no success. I started evaluating Asterisk a few months ago, i also tried with such simple configurations and did not have issues like this. Here is my Asterisk version: Asterisk CVS-02/21/04-16:21:31 built by root@localhost.localdomain on a i686 running Linux I am really curious if you were able to solve the problem. If so, what was the reason behind that weird behaviour and how did you solve it? If not, can anyone please tell what is going wrong? Regards, Girish BTW, J Poz, dont use reinvite=, it does not exist, use canreinvite= instead.>From: J Poz >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing >Date: Sun, 2 May 2004 16:41:52 -0700 (PDT) > >Sorry for any confusion.........But in my latest error, instead of calling >my clients "jay" and "jtest", I'm calling them "400" and "410".. Everything >else is still the same and it's same problem. > >My guess is that I've set a parameter incorrectly and therefore Asterisk >thinks there's only one client so any calls I try to make between the two >fail since it thinks the other client is busy. But I don't understand >enough to interpret the error message. I thought the SIP part would be the >easy part - I already have the FXO and FXS interfaces working. > >Again, thanks for anyone who can help me since I am at a loss! > >J Poz wrote: >Can anyone help. I've changed the extensions.conf file as follows: > >extensions.conf >[sip] ; context for X-Lite Clients >exten =>11,1,Dial(SIP/jay,20,tr) >exten =>22,1,Dial(SIP/jtest,20,tr) > >I'm still getting the Auto-congesting error (and circuit-busy). Does anyone >know what is causing this in such a simple configuration? > > >localhost*CLI> > -- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in new stack > -- Called 410 >May 2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest: >Auto-congesting SIP/410-a4a1 > -- SIP/410-a4a1 is circuit-busy > == Everyone is busy at this time >_________________________________________________________________ Send flowers in 24 hours! http://www.fabmall.com/affiliatehtml/redir/nl7.asp At MSN Shopping. --------------------------------- Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs --------------------------------- Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040505/81e1b87b/attachment.htm
Girish and All, I got my SIMPLE SIMPLE X-Lite to X-Lite configuration to work. First, I want to thank "William Ray" who helped me offline. I found him by one of his postings in the http://asterisk.xvoip.com/ forum where he helped someone else with a similar issue. He gave me a working configuration for what I was trying to do and it worked. However, I still needed to debug and find out what was wrong with my configuration since I know it can help someone else in the future (if they search the mailing list for the same or similar problem). The main problem that PREVENTED the X-Lite softphones from communicating with eachother was I had invalid syntax in the sip.conf file for the caller-id field as follows: Incorrect Syntax: callerid="Jay <400>" Correct Syntax: callerid="Jay" <400> It blows my mind why asterisk would spit out "Auto-congesting" and "is circuit-busy" errors since I don't know how they're related. That was the main problem that needed to get fixed for the configuration to work. Also, I then changed the DTMF mode to "rfc2833" from "inband" to get rid of "Unable to process inband DTMF on 2 frames" warnings. Everything is PEACHY now.. I hope this proves useful to others in the future. I invested over 30 hours on this problem so hopefully it can be avoided by someone else. J.......... J Poz <jpoz0000@yahoo.com> wrote: Girish, Thanks for replying and trying to work my "simple configuration". Nobody on the list has replied with any help and I still have the problem. I've invested well over 20 hours on this problem and still don't have a solution (I have everything else within Asterisk working including IVR menus, X100 interfaces, etc). However, I am not able to get a simple Softphone to Softphone configuration to work. Can anyone on the LIST help us Girish Gopinath <gopinath_girish@hotmail.com> wrote: Hello, Replying to the mail which was posted 3 days back. I tested the configuration here with SJphones, and got the same error: "circuit-busy". I tried with sip debug turned on, and found that asterisk receives a CANCEL request from the user agent immediately after it receives INVITE. When i first saw this mail, i thought it was a simple config issue, but even after trying for more than 2 hours, i am not able to figure out why it is happening. I tried changing the sip.conf entries with the minimum required values, but no success. I started evaluating Asterisk a few months ago, i also tried with such simple configurations and did not have issues like this. Here is my Asterisk version: Asterisk CVS-02/21/04-16:21:31 built by root@localhost.localdomain on a i686 running Linux I am really curious if you were able to solve the problem. If so, what was the reason behind that weird behaviour and how did you solve it? If not, can anyone please tell what is going wrong? Regards, Girish BTW, J Poz, dont use reinvite=, it does not exist, use canreinvite= instead.>From: J Poz >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing >Date: Sun, 2 May 2004 16:41:52 -0700 (PDT) > >Sorry for any confusion.........But in my latest error, instead of calling >my clients "jay" and "jtest", I'm calling them "400" and "410".. Everything >else is still the same and it's same problem. > >My guess is that I've set a parameter incorrectly and therefore Asterisk >thinks there's only one client so any calls I try to make between the two >fail since it thinks the other client is busy. But I don't understand >enough to interpret the error message. I thought the SIP part would be the >easy part - I already have the FXO and FXS interfaces working. > >Again, thanks for anyone who can help me since I am at a loss! > >J Poz wrote: >Can anyone help. I've changed the extensions.conf file as follows: > >extensions.conf >[sip] ; context for X-Lite Clients >exten =>11,1,Dial(SIP/jay,20,tr) >exten =>22,1,Dial(SIP/jtest,20,tr) > >I'm still getting the Auto-congesting error (and circuit-busy). Does anyone >know what is causing this in such a simple configuration? > > >localhost*CLI> > -- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in new stack > -- Called 410 >May 2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest: >Auto-congesting SIP/410-a4a1 > -- SIP/410-a4a1 is circuit-busy > == Everyone is busy at this time >_________________________________________________________________ Send flowers in 24 hours! http://www.fabmall.com/affiliatehtml/redir/nl7.asp At MSN Shopping. --------------------------------- Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs --------------------------------- Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040505/0a2b89b8/attachment.htm
Paul Tyreman
2004-May-06 01:12 UTC
[Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing
Has anyone managed to get a stutter tone working on the X-Lite clients when that extension has voicemail ?? -------------------------------------------------------------------------------- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of J Poz Posted At: 05 May 2004 22:46 Posted To: Asterisk-Users Conversation: [Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing Subject: Re: [Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing Girish and All, I got my SIMPLE SIMPLE X-Lite to X-Lite configuration to work. First, I want to thank "William Ray" who helped me offline. I found him by one of his postings in the http://asterisk.xvoip.com/ forum where he helped someone else with a similar issue. He gave me a working configuration for what I was trying to do and it worked. However, I still needed to debug and find out what was wrong with my configuration since I know it can help someone else in the future (if they search the mailing list for the same or similar problem). The main problem that PREVENTED the X-Lite softphones from communicating with eachother was I had invalid syntax in the sip.conf file for the caller-id field as follows: Incorrect Syntax: callerid="Jay <400>" Correct Syntax: callerid="Jay" <400> It blows my mind why asterisk would spit out "Auto-congesting" and "is circuit-busy" errors since I don't know how they're related. That was the main problem that needed to get fixed for the configuration to work. Also, I then changed the DTMF mode to "rfc2833" from "inband" to get rid of "Unable to process inband DTMF on 2 frames" warnings. Everything is PEACHY now.. I hope this proves useful to others in the future. I invested over 30 hours on this problem so hopefully it can be avoided by someone else. J.......... J Poz <jpoz0000@yahoo.com> wrote: Girish, Thanks for replying and trying to work my "simple configuration". Nobody on the list has replied with any help and I still have the problem. I've invested well over 20 hours on this problem and still don't have a solution (I have everything else within Asterisk working including IVR menus, X100 interfaces, etc). However, I am not able to get a simple Softphone to Softphone configuration to work. Can anyone on the LIST help us Girish Gopinath <gopinath_girish@hotmail.com> wrote: Hello, Replying to the mail which was posted 3 days back. I tested the configuration here with SJphones, and got the same error: "circuit-busy". I tried with sip debug turned on, and found that asterisk receives a CANCEL request from the user agent immediately after it receives INVITE. When i first saw this mail, i thought it was a simple config issue, but even after trying for more than 2 hours, i am not able to figure out why it is happening. I tried changing the sip.conf entries with the minimum required values, but no success. I started evaluating Asterisk a few months ago, i also tried with such simple configurations and did not have issues like this. Here is my Asterisk version: Asterisk CVS-02/21/04-16:21:31 built by root@localhost.localdomain on a i686 running Linux I am really curious if you were able to solve the problem. If so, what was the reason behind that weird behaviour and how did you solve it? If not, can anyone please tell what is going wrong? Regards, Girish BTW, J Poz, dont use reinvite=, it does not exist, use canreinvite= instead.>From: J Poz >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing >Date: Sun, 2 May 2004 16:41:52 -0700 (PDT) > >Sorry for any confusion.........But in my latest error, instead of calling >my clients "jay" and "jtest", I'm calling them "400" and "410".. Everything >else is still the same and it's same problem. > >My guess is that I've set a parameter incorrectly and therefore Asterisk >thinks there's only one client so any calls I try to make between the two >fail since it thinks the other client is busy. But I do n't understand >enough to interpret the error message. I thought the SIP part would be the >easy part - I already have the FXO and FXS interfaces working. > >Again, thanks for anyone who can help me since I am at a loss! > >J Poz wrote: >Can anyone help. I've changed the extensions.conf file as follows: > >extensions.conf >[sip] ; context for X-Lite Clients >exten =>11,1,Dial(SIP/jay,20,tr) >exten =>22,1,Dial(SIP/jtest,20,tr) > >I'm still getting the Auto-congesting error (and circuit-busy). Does anyone >know what is causing this in such a simple configuration? > > >localhost*CLI> > -- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in new stack > -- Called 410 >May 2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest: >Auto-congesting SIP/410-a4a1 > -- SIP/410-a4a1 is circuit-busy > == Everyone is busy at this time >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040506/0936768a/attachment.htm