Re: [asterisk-users] All trunk are busy please try your call again later
Good Day Find attached the relevant portions of the asterisk CLI. Please,which portion of the extension .conf should i send ? It is connected via RJ 45 connector to an E1 modem to the telco company. I use E1 link. I will appreciate your reply. Best Regards On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] wrote: lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users SIP SHOW PEERS Name/username HostDyn Nat ACL Port Status 7871/7871 (Unspecified)D 0Unmonitored ... ... 7874/7874 (Unspecified)D 0Unmonitored 108 sip peers [108 online , 0 offline] Verbosity is at least 3 ZAP SHOW CHANNELS Chan Extension Context Language MusicOnHold pseudodefault en 1default en 2default en ZAP SHOW CHANNELS Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 Verbosity is at least 3 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
Thanks Please am using putty to again access to my Linux asterisk box. How can i use tcpdump to get your request on the exact Ethernet port and port number. I will appreciate your reply. Best Regards On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] wrote: lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
Hi Steve Am connected to the telco through an E1 link using modem(Watson 5 modem SDHSL 1 PAIR schmid telecommunications).The MODEM is connected to the asterisk box through RJ 45 to the asterisk box end and serial connector to the modem end . Which portion of the extension conf should i post ? Thanks On Dec 18, 2007 12:03 PM, Steve Totaro [EMAIL PROTECTED] wrote: Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success You need to at least post some verbose from the console and explain how you are connecting to the PSTN. It would greatly help if you included the relevant portions of your extensions.conf. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
lolu,
while you are making the call., capture and post your CLI output
... this is easy to do since you are using putty.
login to your pbx and start asterisk, use the below command:
# asterisk -vvvr
then make the call. hilite the text on the putty terminal and paste it
into the body of the email to the list...
sorry if I'm making these instruction too basic...
pbv01*CLI
-- Executing [EMAIL PROTECTED]:1] Wait("SIP/202-b753da18", "1") in new
stack
-- Executing [EMAIL PROTECTED]:2] Answer("SIP/202-b753da18", "") in
new stack
-- Executing [EMAIL PROTECTED]:3] NoOp("SIP/202-b753da18", "DEBUG:
CALLERID=") in new stack
-- Executing [EMAIL PROTECTED]:4] Notify("SIP/202-b753da18",
"800202|x202|300/192.168.15.100") in new stack
-- Notify: sending '800202|x202|300' to 192.168.15.100:4
-- Executing [EMAIL PROTECTED]:5] AGI("SIP/202-b753da18",
"agi-callpop4.sh||red") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-callpop4.sh
-- AGI Script agi-callpop4.sh completed, returning 0
-- Executing [EMAIL PROTECTED]:6] NoOp("SIP/202-b753da18", "AGISTATUS
is FAILURE") in new stack
-- Executing [EMAIL PROTECTED]:7] NoOp("SIP/202-b753da18", "DEBUG:
EXTEN=300") in new stack
-- Executing [EMAIL PROTECTED]:8] Dial("SIP/202-b753da18",
"SIP/300|15|rt") in new stack
-- Called 300
-- SIP/300-09e062e8 is ringing
== Spawn extension (local-sip, 300, 8) exited non-zero on
'SIP/202-b753da18'
daveC
Lolu Gbenga wrote:
Thanks
Please am using putty to again access to my Linux asterisk box.
How can i use tcpdump to get your request on the exact Ethernet port
and port number.
I will appreciate your reply.
Best Regards
On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED]
wrote:
lolu,
sounds more like a telco/itsp problem then *.
I would
tcpdump -i eth0 port 5060
to make sure it is actually going out... change 5060 if you have changed
your port to your itsp, of course.
to see what is going on as well as the other debugging notes mentioned
in this thread.
daveC
Lolu Gbenga wrote:
Good Day all
Please I am having some issues on my voip asterisk server
I make internal calls on extensions configured ie extension 192
can
call extension 195 etc
But each time i try to make calls outside the extension ie calling
a
GSM or an external line ,i always hear this response "all trunk
calls
are busy please try your call again later"
Please how can i resolve this problem .
I will appreciate your response.
Best Regards
Success
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
My wife's sister is in California.
I should buy her a Videophone2008!
Truly, The Next Best Thing to Being There!
--
WorldWideVideoPhones.com
856.380.0894
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.503 / Virus Database: 269.17.2 - Release Date: 12/14/2007 12:00 AM
--
My wife's sister is in California.
I should buy her a Videophone2008!
Truly, The Next Best Thing to Being There!
--
WorldWideVideoPhones.com
856.380.0894
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
What is the output of ztconfig from the Linux command line? What does your zaptel.conf and zapata.conf look like? What is the relevant part of extensions.conf (the dialout section that fails). Also from the CLI, it would be most helpful to post the output you get when dialing out fails. I don't think it is a network issue at all, I think your configs need some work. Thanks, Steve Totaro Lolu Gbenga wrote: Good Day Find attached the relevant portions of the asterisk CLI. Please,which portion of the extension .conf should i send ? It is connected via RJ 45 connector to an E1 modem to the telco company. I use E1 link. I will appreciate your reply. Best Regards On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
Hi all, I am grateful for our contribution so far . I followed dave advise and i have the attached file using the aterisk -r when a call is made. I attached two files. One of the attached file is for the external call,which replied with the PROBLEM all trunks are busy now,please try your call again later. The second attachment is when i made internal calls and the phone rang. Please,i will be expecting your replies for further directions. Best Regards On Dec 20, 2007 2:58 PM, Steve Totaro [EMAIL PROTECTED] wrote: What is the output of ztconfig from the Linux command line? What does your zaptel.conf and zapata.conf look like? What is the relevant part of extensions.conf (the dialout section that fails). Also from the CLI, it would be most helpful to post the output you get when dialing out fails. I don't think it is a network issue at all, I think your configs need some work. Thanks, Steve Totaro Lolu Gbenga wrote: Good Day Find attached the relevant portions of the asterisk CLI. Please,which portion of the extension .conf should i send ? It is connected via RJ 45 connector to an E1 modem to the telco company. I use E1 link. I will appreciate your reply. Best Regards On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
Hi All I FOUND OUT THAT THE ATTACHMENT WAS NOT SENT WITH THE MAIL. FIND BELOW THE OUTPUT USING asterisk -vvvr command for EXTERNAL calls that gave the ouput ALL TRUNKS ARE BUSY PLEASE TRY YOUR CALL LATER. Verbosity is at least 3 -- Executing Macro(SIP/7871-f813, dialout-trunk|1|018774957||) in new sta ck -- Executing GotoIf(SIP/7871-f813, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/7871-f813, user-callerid) in new stack -- Executing Set(SIP/7871-f813, AMPUSER=7871) in new stack -- Executing Set(SIP/7871-f813, EMERGENCYCID=7871) in new stack -- Executing Set(SIP/7871-f813, AMPUSERCIDNAME=7871) in new stack -- Executing GotoIf(SIP/7871-f813, 0?6) in new stack -- Executing Set(SIP/7871-f813, CALLERID(all)=7871 7871) in new stack -- Executing NoOp(SIP/7871-f813, Using CallerID 7871 7871) in new stack -- Executing Macro(SIP/7871-f813, record-enable|7871|OUT) in new stack -- Executing GotoIf(SIP/7871-f813, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/7871-f813, recordingcheck|20051006-001624|1128554184. 8) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20051006-001624|1128554184.8: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/7871-f813, No recording needed) in new stack -- Executing Macro(SIP/7871-f813, outbound-callerid|1) in new stack -- Executing Set(SIP/7871-f813, USEROUTCID=7871) in new stack -- Executing GotoIf(SIP/7871-f813, 1?4) in new stack -- Goto (macro-outbound-callerid,s,4) -- Executing GotoIf(SIP/7871-f813, 0?6) in new stack -- Executing Set(SIP/7871-f813, CALLERID(all)=7871) in new stack -- Executing GotoIf(SIP/7871-f813, 1?8) in new stack -- Goto (macro-outbound-callerid,s,8) -- Executing NoOp(SIP/7871-f813, CallerID set to 7871) in new stack -- Executing Set(SIP/7871-f813, GROUP()=OUT_1) in new stack -- Executing GotoIf(SIP/7871-f813, 0?108) in new stack -- Executing Set(SIP/7871-f813, DIAL_NUMBER=018774957) in new stack -- Executing Set(SIP/7871-f813, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/7871-f813, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Removed prefix. New number: 8774957 -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/7871-f813, OUTNUM=8774957) in new stack -- Executing Set(SIP/7871-f813, custom=ZAP/1) in new stack -- Executing GotoIf(SIP/7871-f813, 0?16) in new stack -- Executing Dial(SIP/7871-f813, ZAP/1/8774957|120|W) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1/8774957 -- Zap/1-1 is proceeding passing it to SIP/7871-f813 Don't know what to do if second ROSE component is of type 0x6 -- Channel 0/1, span 1 got hangup request -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(SIP/7871-f813, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/7871-f813, Dial failed due to CHANUNAVAIL) in new s tack -- Executing Macro(SIP/7871-f813, outisbusy|) in new stack -- Executing Playback(SIP/7871-f813, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/7871-f813, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'en') -- Executing Macro(SIP/7871-f813, hangupcall) in new stack -- Executing ResetCDR(SIP/7871-f813, w) in new stack -- Executing NoCDR(SIP/7871-f813, ) in new stack -- Executing Wait(SIP/7871-f813, 5) in new stack -- Executing Hangup(SIP/7871-f813, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/7871-f813' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/7871-f813' in macro 'outisbusy' == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/7871-f813' asterisk1*CLI ALSO FIND BELOW THE OUTPUT using asterisk -vvvr command FOR INTERNAL calls that rang. Verbosity is at least 3 -- Executing Macro(SIP/7871-bb64, exten-vm|novm|7874) in new stack -- Executing Macro(SIP/7871-bb64, user-callerid) in new stack -- Executing Set(SIP/7871-bb64, AMPUSER=7871) in new stack -- Executing Set(SIP/7871-bb64, EMERGENCYCID=7871) in new stack -- Executing Set(SIP/7871-bb64, AMPUSERCIDNAME=7871) in new stack -- Executing GotoIf(SIP/7871-bb64, 0?6) in new stack -- Executing Set(SIP/7871-bb64, CALLERID(all)=7871 7871) in new stack -- Executing NoOp(SIP/7871-bb64, Using CallerID
Re: [asterisk-users] All trunk are busy please try your call again later
lolu
I reformated the output so it was easier to understand. I attached the
word document for you.
on the below line:
--
Executing Dial("SIP/7871-f813", "ZAP/1/8774957|120|W") in
new stack
-- Requested transfer capability:
0x00 - SPEECH
-- Called 1/8774957
-- Zap/1-1 is proceeding passing it
to SIP/7871-f813 Don't know what to do if second ROSE component is
of type 0x6
it looks
like this is where it determines it can't proceed... also, there are
many tests along the way... we don't know about the
questions/conditions and if that effects it or not... probably not..
in any case, the question you must answer is 'what is the second
ROSE component'??? and why is of type 0x6???
how is it set and by what component?
hope that moves you closer to the ultimate resolution...
daveC
Lolu Gbenga wrote:
Hi All
I FOUND OUT THAT THE ATTACHMENT WAS NOT SENT WITH THE MAIL.
FIND BELOW THE OUTPUT USING asterisk -vvvr command for EXTERNAL calls
that gave the ouput ALL TRUNKS ARE BUSY PLEASE TRY YOUR CALL LATER.
Verbosity is at least 3
-- Executing Macro("SIP/7871-f813", "dialout-trunk|1|018774957||")
in new sta ck
-- Executing GotoIf("SIP/7871-f813", "1?3:2") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/7871-f813", "user-callerid") in new stack
-- Executing Set("SIP/7871-f813", "AMPUSER=7871") in new stack
-- Executing Set("SIP/7871-f813", "EMERGENCYCID=7871") in new stack
-- Executing Set("SIP/7871-f813", "AMPUSERCIDNAME=7871") in new
stack
-- Executing GotoIf("SIP/7871-f813", "0?6") in new stack
-- Executing Set("SIP/7871-f813", "CALLERID(all)="7871" 7871") in
new stack
-- Executing NoOp("SIP/7871-f813", "Using CallerID "7871" 7871")
in new stack
-- Executing Macro("SIP/7871-f813", "record-enable|7871|OUT") in
new stack
-- Executing GotoIf("SIP/7871-f813", "0 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/7871-f813",
"recordingcheck|20051006-001624|1128554184. 8") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20051006-001624|1128554184.8: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/7871-f813", "No recording needed") in new
stack
-- Executing Macro("SIP/7871-f813", "outbound-callerid|1") in new
stack
-- Executing Set("SIP/7871-f813", "USEROUTCID=7871") in new stack
-- Executing GotoIf("SIP/7871-f813", "1?4") in new stack
-- Goto (macro-outbound-callerid,s,4)
-- Executing GotoIf("SIP/7871-f813", "0?6") in new stack
-- Executing Set("SIP/7871-f813", "CALLERID(all)=7871") in new
stack
-- Executing GotoIf("SIP/7871-f813", "1?8") in new stack
-- Goto (macro-outbound-callerid,s,8)
-- Executing NoOp("SIP/7871-f813", "CallerID set to "" 7871") in
new stack
-- Executing Set("SIP/7871-f813", "GROUP()=OUT_1") in new stack
-- Executing GotoIf("SIP/7871-f813", "0?108") in new stack
-- Executing Set("SIP/7871-f813", "DIAL_NUMBER=018774957") in new
stack
-- Executing Set("SIP/7871-f813", "DIAL_TRUNK=1") in new stack
-- Executing AGI("SIP/7871-f813", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Removed prefix. New number: 8774957
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/7871-f813", "OUTNUM=8774957") in new stack
-- Executing Set("SIP/7871-f813", "custom=ZAP/1") in new stack
-- Executing GotoIf("SIP/7871-f813", "0?16") in new stack
-- Executing Dial("SIP/7871-f813", "ZAP/1/8774957|120|W") in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1/8774957
-- Zap/1-1 is proceeding passing it to SIP/7871-f813
Don't know what to do if second ROSE component is of type 0x6
-- Channel 0/1, span 1 got hangup request
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/7871-f813", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/7871-f813", "Dial failed due to
CHANUNAVAIL") in new s tack
-- Executing Macro("SIP/7871-f813", "outisbusy|") in new stack
-- Executing Playback("SIP/7871-f813", "all-circuits-busy-now") in
new stack
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/7871-f813", "pls-try-call-later") in new
stack
-- Playing 'pls-try-call-later' (language 'en')
-- Executing Macro("SIP/7871-f813", "hangupcall") in new stack
-- Executing ResetCDR("SIP/7871-f813", "w") in new stack
-- Executing NoCDR("SIP/7871-f813", "") in new stack
-- Executing Wait("SIP/7871-f813", "5") in new stack
-- Executing Hangup("SIP/7871-f813", "") in new stack
== Spawn
[asterisk-users] All trunk are busy please try your call again later
Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
Post: Asterisk CLI : sip show peers Asterisk CLI : zap show channels Asterisk CLI: zap show status As well as your extensions.conf Are you able to ping you GSM gateway? is connected via SIP or Telephony interface card? Best regards, Mouta On Dec 18, 2007 10:47 AM, Lolu Gbenga [EMAIL PROTECTED] wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success You need to at least post some verbose from the console and explain how you are connecting to the PSTN. It would greatly help if you included the relevant portions of your extensions.conf. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
