[asterisk-users] Handoff dial control to dialplan after AMI Originate
Hello,
I'd like to use the AMI interface to originate a call to a context in a
dialplan, and handoff the dial control to the context.
Whenever I execute the below action, the recipient does ring, but when I answer
it dials the recipient again. I believe this is because once answered the
system is going to execute the Context/Exten/Prio in the Originate action?
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialContext
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
Anyone have an idea how to fix this?
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Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate
Looks correct to me
2013/6/19 Grant Bagdasarian g...@cm.nl
Hello,
** **
I’d like to use the AMI interface to originate a call to a context in a
dialplan, and handoff the dial control to the context.
** **
Whenever I execute the below action, the recipient does ring, but when I
answer it dials the recipient again. I believe this is because once
answered the system is going to execute the Context/Exten/Prio in the
Originate action?
** **
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialContext
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
** **
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
** **
Anyone have an idea how to fix this?
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Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate
I fixed it. The problem is just as I assumed; once the call is answered the
dialplan goes into what’s defined in Context/Exten/Prio of the Originate action.
I changed the Context/Exten/Prio in the action and pointed it to something
else. Now it works.
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialProcessor
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
[originateDialProcessor]
exten = outbound1,1,Wait(1)
exten = outbound1,n,NoOp(${DIALSTATUS})
exten = outbound1,n,Hangup
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, June 19, 2013 10:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI
Originate
Looks correct to me
2013/6/19 Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl
Hello,
I’d like to use the AMI interface to originate a call to a context in a
dialplan, and handoff the dial control to the context.
Whenever I execute the below action, the recipient does ring, but when I answer
it dials the recipient again. I believe this is because once answered the
system is going to execute the Context/Exten/Prio in the Originate action?
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialContext
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
Anyone have an idea how to fix this?
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Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate
Why can’t I execute any more dialplan after the Dial application? The scenario
is when the Dial application dials the recipient but the recipient doesn’t
answer. The AMI will never go into the originateDialProcessor because the call
was never answered. So I expect the Dialplan to continue after the Dial
application has reached its timeout.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: Wednesday, June 19, 2013 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI
Originate
I fixed it. The problem is just as I assumed; once the call is answered the
dialplan goes into what’s defined in Context/Exten/Prio of the Originate action.
I changed the Context/Exten/Prio in the action and pointed it to something
else. Now it works.
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialProcessor
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
[originateDialProcessor]
exten = outbound1,1,Wait(1)
exten = outbound1,n,NoOp(${DIALSTATUS})
exten = outbound1,n,Hangup
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, June 19, 2013 10:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI
Originate
Looks correct to me
2013/6/19 Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl
Hello,
I’d like to use the AMI interface to originate a call to a context in a
dialplan, and handoff the dial control to the context.
Whenever I execute the below action, the recipient does ring, but when I answer
it dials the recipient again. I believe this is because once answered the
system is going to execute the Context/Exten/Prio in the Originate action?
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialContext
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
Anyone have an idea how to fix this?
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Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate
Did you specify a timeout value? jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate
On Wed, Jun 19, 2013 at 4:00 PM, Grant Bagdasarian g...@cm.nl wrote:
Why can’t I execute any more dialplan after the Dial application? The
scenario is when the Dial application dials the recipient but the recipient
doesn’t answer. The AMI will never go into the originateDialProcessor
because the call was never answered. So I expect the Dialplan to continue
after the Dial application has reached its timeout.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Grant Bagdasarian
*Sent:* Wednesday, June 19, 2013 11:24 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Handoff dial control to dialplan after
AMI Originate
** **
I fixed it. The problem is just as I assumed; once the call is answered
the dialplan goes into what’s defined in Context/Exten/Prio of the
Originate action.
** **
I changed the Context/Exten/Prio in the action and pointed it to something
else. Now it works.
** **
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialProcessor
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
** **
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
** **
[originateDialProcessor]
exten = outbound1,1,Wait(1)
exten = outbound1,n,NoOp(${DIALSTATUS})
exten = outbound1,n,Hangup
** **
*From:* asterisk-users-boun...@lists.digium.com [
mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com]
*On Behalf Of *Lenz Emilitri
*Sent:* Wednesday, June 19, 2013 10:10 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Handoff dial control to dialplan after
AMI Originate
** **
Looks correct to me
** **
2013/6/19 Grant Bagdasarian g...@cm.nl
Hello,
I’d like to use the AMI interface to originate a call to a context in a
dialplan, and handoff the dial control to the context.
Whenever I execute the below action, the recipient does ring, but when I
answer it dials the recipient again. I believe this is because once
answered the system is going to execute the Context/Exten/Prio in the
Originate action?
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialContext
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
Anyone have an idea how to fix this?
--
You need a special extension 'failed' in a context originateDialProcessor
to catch the control when call doesn't get answered in first leg.
[originateDialProcessor]
exten = outbound1,1,Wait(1)
exten = outbound1,n,NoOp(${DIALSTATUS})
exten = outbound1,n,Hangup
exten = failed,1,NoOp(- CALL DIDN'T GET ANSWERED IN FIRST LEG -)
--Satish Barot
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Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate
I used a combination of failed and the h extension to get the dialplan to do
what I want.
Thanks for the help guys!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Sent: Wednesday, June 19, 2013 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI
Originate
On Wed, Jun 19, 2013 at 4:00 PM, Grant Bagdasarian
g...@cm.nlmailto:g...@cm.nl wrote:
Why can't I execute any more dialplan after the Dial application? The scenario
is when the Dial application dials the recipient but the recipient doesn't
answer. The AMI will never go into the originateDialProcessor because the call
was never answered. So I expect the Dialplan to continue after the Dial
application has reached its timeout.
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Grant Bagdasarian
Sent: Wednesday, June 19, 2013 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI
Originate
I fixed it. The problem is just as I assumed; once the call is answered the
dialplan goes into what's defined in Context/Exten/Prio of the Originate action.
I changed the Context/Exten/Prio in the action and pointed it to something
else. Now it works.
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialProcessor
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
[originateDialProcessor]
exten = outbound1,1,Wait(1)
exten = outbound1,n,NoOp(${DIALSTATUS})
exten = outbound1,n,Hangup
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, June 19, 2013 10:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI
Originate
Looks correct to me
2013/6/19 Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl
Hello,
I'd like to use the AMI interface to originate a call to a context in a
dialplan, and handoff the dial control to the context.
Whenever I execute the below action, the recipient does ring, but when I answer
it dials the recipient again. I believe this is because once answered the
system is going to execute the Context/Exten/Prio in the Originate action?
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialContext
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
Anyone have an idea how to fix this?
--
You need a special extension 'failed' in a context originateDialProcessor to
catch the control when call doesn't get answered in first leg.
[originateDialProcessor]
exten = outbound1,1,Wait(1)
exten = outbound1,n,NoOp(${DIALSTATUS})
exten = outbound1,n,Hangup
exten = failed,1,NoOp(- CALL DIDN'T GET ANSWERED IN FIRST LEG -)
--Satish Barot
--
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Re: [asterisk-users] Queue Limit Callers
Thanks all for the inputs... Let me work on it and come back again with some results... From: Ioan Indreias indre...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 18, 2013 1:43 PM Subject: Re: [asterisk-users] Queue Limit Callers Hello Shanavaz., Please find some quick thoughts: * 2 main queues * agents logged on one or on both main queues * before sending a new call to one of the main queues check the number of waiting callers (QUEUE_WAITING_COUNT function) and divert (for example for 30 sec) the call on a empty members queue/parking slot/music-on-hold if the queue threshold is reached. The threshold could be read from a database, internal astdb or could be set as a global variable updated when agents login/logout/pause/unpause or could be dynamically computed based on QUEUE_MEMBER_COUNT / QUEUE_MEMBER_LIST After the divert period is ended the call will return and the threshold is checked again, etc. This method have some negative impacts (the entry position number for calls over the threshold //origposition// will have no meaning, a newer call could be served before an older one, etc.) but you could manipulate the call flow exactly how you want. HTH, Ioan http://www.modulo.ro On Tue, Jun 18, 2013 at 12:05 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: You should have different sets of agents logged in to different queues and you should have a monitor to move them from one queue to the other based on incoming traffic. l. 2013/6/17 Shanavaz E A shanava...@yahoo.com Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I have two Queues - Sales Booking I have 12 Agents who are added to both the queues Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales Queue. Only 8 calls in the Booking Queue should hit the Agents and the other 4 calls should remain in hold. 4 calls in the Sales Queue should hit the other 4 agents and the other 2 call should be in hold. Means at a time a maximum of 8 Booking calls only should hit the agents and 4 Sales Calls only should hit the agents. If number of logged in agents are less, proportionally the number of call limit should be reduced. For example, if there are only 10 agents, 7 Booking Calls should hit and 3 Sales calls should hit. The idea is that all agents should be able to answer calls in both queues in rotation. Otherwise its possible to add some agents to booking queue and other agents to sales queue. But thats not what is required. Kindly help if there is some idea to implement this. RegardsShanavaz. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate
Your both channels legs are identical strings. It should be like this.
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialContext2
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678,callerid1=00311234567
Timeout: 1
** **
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
[originateDialContext2]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Dial(SIP/${callerid1}@originateChannel)
On Wed, Jun 19, 2013 at 11:20 AM, Grant Bagdasarian g...@cm.nl wrote:
Hello,
** **
I’d like to use the AMI interface to originate a call to a context in a
dialplan, and handoff the dial control to the context.
** **
Whenever I execute the below action, the recipient does ring, but when I
answer it dials the recipient again. I believe this is because once
answered the system is going to execute the Context/Exten/Prio in the
Originate action?
** **
Action: Originate
Channel: Local/outbound1@originateDialContext
CallerID: 00311234567
Context: originateDialContext
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 1
** **
[originateDialContext]
exten = outbound1,1,Wait(1)
exten = outbound1,n,Set(recipient=${recipient})
exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)
** **
Anyone have an idea how to fix this?
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[asterisk-users] SIP Simple support on Asterisk 11
Hi all, I am trying to enable SIP SIMPLE communication in my test environment. I have the following env : - one server (192.168.50.126) with Asterisk 11 - 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143 I successfully had a phone call between clients. I used the following link to enable SIMPLE messaging between my clients : http://highsecurity.blogspot.ca/2012/03/asterisk-10-110-sms-messaging-or-sip.html Both users managed to register. Adding verbose on the server, I have the following traces when I send the message MESSAGE FROM ALICE TO BOB from demo-alice to demo-bob http://paste.fedoraproject.org/19489/37158861/ As you can see I succeed to have the message sent from alice to Asterisk. When the server is trying to transmitting, I see a 401 error message. According to this post (http://forums.digium.com/viewtopic.php?f=1t=72814) the first 401 should be normal as authentication is requested. Afterwards the server emit 202 message. But demo-bob never receives a message. I ran wireshark on server and client. It confirms that no message is sent from Asterisk to demo-bob. Could you please give me advice ? Here are my extensions.conf and sip.conf according to the link I mentioned. http://paste.fedoraproject.org/19626/16493741/ http://paste.fedoraproject.org/19627/49423137/ Thanks a lot, Eloi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PCI Passthrough of T1 cards
On Jun 16, 2013, at 4:27 PM, Nick Khamis sym...@gmail.com wrote: Anyone try this? I saw a post here: http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/94041-setup-of-sangoma-a101-in-my-elastix.html But not sure if it's possible. What I am asking is if there are any T1 cards with virtual functions implemented in their drivers to allow pci-passthrough? PCI pass through is a function of the virtual machine's host system, not with the t1 card drivers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PCI Passthrough of T1 cards
Hello James, Thank you so much for your response. I should have chose my words carefully. PCI pass-through in terms of virtualization of devices and it's draw back are well know. I was leaning more towards near host performance virtualization using SR-IOV. This moves emphasis back to the production drivers of the interface card using virtual functions etc., and can provide near host performance. Rephrasing my question, are any of the T1 pci manufactures providing support for virtualization using SR-IOV and virutal functions? Kind Regards, Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Simple support on Asterisk 11
Eloi Bail wrote:
I am trying to enable SIP SIMPLE communication in my test environment.
I have the following env :
- one server (192.168.50.126) with Asterisk 11
- 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143
I successfully had a phone call between clients.
I used the following link to enable SIMPLE messaging between my clients :
http://highsecurity.blogspot.ca/2012/03/asterisk-10-110-sms-messaging-or-sip.html
Both users managed to register.
Adding verbose on the server, I have the following traces when I send the
message MESSAGE FROM ALICE TO BOB from demo-alice to demo-bob
http://paste.fedoraproject.org/19489/37158861/
As you can see I succeed to have the message sent from alice to Asterisk.
When the server is trying to transmitting, I see a 401 error message.
According to this post ( http://forums.digium.com/viewtopic.php?f=1t=72814 )
the first 401 should be normal as authentication is requested.
Afterwards the server emit 202 message.
But demo-bob never receives a message.
I ran wireshark on server and client. It confirms that no message is sent from
Asterisk to demo-bob.
Could you please give me advice ?
Here are my extensions.conf and sip.conf according to the link I mentioned.
http://paste.fedoraproject.org/19626/16493741/
http://paste.fedoraproject.org/19627/49423137/
Eloi,
The trace shows that the initial MESSAGE from Alice does not include an
Authorization header so Asterisk responds with a 401 Unauthorized. Alice then
replies with a MESSAGE with an Authorization header, but reuses the same CSeq
header (CSeq: 6 MESSAGE) which causes Asterisk to ignore it as a retransmit:
[Jun 18 16:49:35] DEBUG[16266] chan_sip.c: Ignoring SIP message because of
retransmit (MESSAGE Seqno 6, ours 6)
I believe this is a bug in Pidgin because RFC 3261 [1] states:
CSeq or Command Sequence contains an integer and a method name. The
CSeq number is incremented for each new request within a dialog and
is a traditional sequence number.
...
Requests within a dialog MUST contain strictly monotonically
increasing and contiguous CSeq sequence numbers (increasing-by-one)
in each direction (excepting ACK and CANCEL of course, whose numbers
equal the requests being acknowledged or cancelled).
However, there is also a similar issue [2] that can be worked around by setting
pedantic=no in sip.conf. If that doesn't work, you can give the following
(untested) patch to chan_sip.c a try:
--- chan_sip.c.orig 2013-06-19 11:44:38.0 -0400
+++ chan_sip.c 2013-06-19 11:47:22.0 -0400
@@ -28078,6 +28078,7 @@
} else if (p-icseq
p-icseq == seqno
req-method != SIP_ACK
+ p-method != SIP_MESSAGE
(p-method != SIP_CANCEL || p-alreadygone)) {
/* ignore means don't do anything with it but still have to
respond appropriately. We do this if we receive a repeat of
Good luck and please let the list know how this works out.
[1] http://www.ietf.org/rfc/rfc3261.txt
[2] https://issues.asterisk.org/jira/browse/ASTERISK-19139
Regards,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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Re: [asterisk-users] SIP Simple support on Asterisk 11
On Wednesday, June 19, 2013 11:11:17 AM Matthew J. Roth wrote: Eloi Bail wrote: I am trying to enable SIP SIMPLE communication in my test environment. I use the following which semi-enables message broadcasting to multiple devices so a user who receives a message can reply from any of the devices. http://messinet.com/trac/wiki/Asterisk/Message -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Simple support on Asterisk 11
Hi,
Thanks a lot for this detailed answer :
- I managed to have it working disabling auth message request
: auth_message_requests = no in sip.conf
- pedantic=no does not resolve the issue
- reenabling auth_message_requests = yes and removing pedantic option,
your patch in chan_sip resolves the issues !
As it looks like pidgin has an issue, I guess that we can use it as a
workaround.
I would like know to enable presence notification between each users. To
fulfill it, I am using
http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265377.html
Am I doing it in a good way ?
Thanks !
Eloi
On Wed, Jun 19, 2013 at 12:11 PM, Matthew J. Roth mr...@imminc.com wrote:
Eloi Bail wrote:
I am trying to enable SIP SIMPLE communication in my test environment.
I have the following env :
- one server (192.168.50.126) with Asterisk 11
- 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143
I successfully had a phone call between clients.
I used the following link to enable SIMPLE messaging between my clients :
http://highsecurity.blogspot.ca/2012/03/asterisk-10-110-sms-messaging-or-sip.html
Both users managed to register.
Adding verbose on the server, I have the following traces when I send the
message MESSAGE FROM ALICE TO BOB from demo-alice to demo-bob
http://paste.fedoraproject.org/19489/37158861/
As you can see I succeed to have the message sent from alice to Asterisk.
When the server is trying to transmitting, I see a 401 error message.
According to this post (
http://forums.digium.com/viewtopic.php?f=1t=72814 )
the first 401 should be normal as authentication is requested.
Afterwards the server emit 202 message.
But demo-bob never receives a message.
I ran wireshark on server and client. It confirms that no message is
sent from
Asterisk to demo-bob.
Could you please give me advice ?
Here are my extensions.conf and sip.conf according to the link I
mentioned.
http://paste.fedoraproject.org/19626/16493741/
http://paste.fedoraproject.org/19627/49423137/
Eloi,
The trace shows that the initial MESSAGE from Alice does not include an
Authorization header so Asterisk responds with a 401 Unauthorized. Alice
then
replies with a MESSAGE with an Authorization header, but reuses the same
CSeq
header (CSeq: 6 MESSAGE) which causes Asterisk to ignore it as a
retransmit:
[Jun 18 16:49:35] DEBUG[16266] chan_sip.c: Ignoring SIP message because
of
retransmit (MESSAGE Seqno 6, ours 6)
I believe this is a bug in Pidgin because RFC 3261 [1] states:
CSeq or Command Sequence contains an integer and a method name. The
CSeq number is incremented for each new request within a dialog and
is a traditional sequence number.
...
Requests within a dialog MUST contain strictly monotonically
increasing and contiguous CSeq sequence numbers (increasing-by-one)
in each direction (excepting ACK and CANCEL of course, whose numbers
equal the requests being acknowledged or cancelled).
However, there is also a similar issue [2] that can be worked around by
setting
pedantic=no in sip.conf. If that doesn't work, you can give the
following
(untested) patch to chan_sip.c a try:
--- chan_sip.c.orig 2013-06-19 11:44:38.0 -0400
+++ chan_sip.c 2013-06-19 11:47:22.0 -0400
@@ -28078,6 +28078,7 @@
} else if (p-icseq
p-icseq == seqno
req-method != SIP_ACK
+ p-method != SIP_MESSAGE
(p-method != SIP_CANCEL || p-alreadygone)) {
/* ignore means don't do anything with it but still have
to
respond appropriately. We do this if we receive a
repeat of
Good luck and please let the list know how this works out.
[1] http://www.ietf.org/rfc/rfc3261.txt
[2] https://issues.asterisk.org/jira/browse/ASTERISK-19139
Regards,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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[asterisk-users] Mailing a fax with mutt does not succeed
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works, but when trying with Asterisk's System command it doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com -a /tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com -a
${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by root.
Any hint will be appreciated.
Elder D. Arohuanca
Lima - Peru
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Re: [asterisk-users] Mailing a fax with mutt does not succeed
Probably Asterisk does not know where mutt is, specify it's path in your System
command.
On 2013-06-19, at 2:03 PM, Daniel - Asterisk earohua...@gmail.com wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works, but when trying with Asterisk's System command it doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com -a /tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com -a
${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by root.
Any hint will be appreciated.
Elder D. Arohuanca
Lima - Peru
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Re: [asterisk-users] Mailing a fax with mutt does not succeed
On Wed, 19 Jun 2013, Daniel - Asterisk wrote:
I'm trying to send a received fax with mutt, when I try it from the
Linux shel it works, but when trying with Asterisk's System command it
doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com -a /tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com -a
${FAXDEST}/${tempfax}.tif)
1) Doesn't mutt expect the body on stdin? (Where's the 'echo' in the
Asterisk command?)
2) Is Asterisk executing as root? Does the Asterisk user ID have read
access to the TIFF?
3) If you use 'verbose()' instead of 'system()' does the command look like
your shell command?
4) Is mutt in the Asterisk user ID's path?
5) If you redirect the output in the system() command to a file, does that
yield any clues? I.e., system(foo /tmp/clue 21)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Mailing a fax with mutt does not succeed
Hi Andre,
I've tried with:
System(echo | /usr/bin/mutt -s New fax earohua...@gmail.com -a
${FAXDEST}/${tempfax})
with no success, value of SYSTEMSTATUS variable is APPERROR
Again it works from Linux shell.
Thanks in advance
Elder
On Wed, Jun 19, 2013 at 1:08 PM, Andre Courchesne voipfor...@gmail.comwrote:
Probably Asterisk does not know where mutt is, specify it's path in your
System command.
On 2013-06-19, at 2:03 PM, Daniel - Asterisk earohua...@gmail.com wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works, but when trying with Asterisk's System command it doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com -a
/tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com -a
${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by
root.
Any hint will be appreciated.
Elder D. Arohuanca
Lima - Peru
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Re: [asterisk-users] Mailing a fax with mutt does not succeed
Why echo | ?
Alsy are you sire of the content of ${FAXDEST} and ${tempfax}.
Add some NoOp before.
On 2013-06-19, at 2:29 PM, Daniel - Asterisk earohua...@gmail.com wrote:
Hi Andre,
I've tried with:
System(echo | /usr/bin/mutt -s New fax earohua...@gmail.com -a
${FAXDEST}/${tempfax})
with no success, value of SYSTEMSTATUS variable is APPERROR
Again it works from Linux shell.
Thanks in advance
Elder
On Wed, Jun 19, 2013 at 1:08 PM, Andre Courchesne voipfor...@gmail.com
wrote:
Probably Asterisk does not know where mutt is, specify it's path in your
System command.
On 2013-06-19, at 2:03 PM, Daniel - Asterisk earohua...@gmail.com wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works, but when trying with Asterisk's System command it doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com -a /tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com -a
${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by root.
Any hint will be appreciated.
Elder D. Arohuanca
Lima - Peru
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Re: [asterisk-users] Mailing a fax with mutt does not succeed
Hi Andre:
I added echo to provide STDIN, I'm sure on variable contents, please see
bellow
Hello Steve,
1. I've just addd echo at my sentence, please see output bellow.
2. Asterisk is executing as root, I think Asterisk has access to read TIF
files since I've used ls, chmod, cp mv from Asterisk's CLI with '!'
character.
3. I don't get you, please give some advice to try using Verbose instead
System
4. I don't know how to get this, but I'm using /usr/bin/mutt as you can see
bellow.
5. I have redirected output of System this way : System(echo |
/usr/bin/mutt -s New fax earohua...@gmail.com -a ${FAXDEST}/${tempfax}
/tmp/ocurrencies.txt 21), ocurrencies.txt is empty.
DIALPLAN:
[ Context 'default' created by 'pbx_config' ]
'*95' = 1. NoOp(trying to send a fax to an email)
2. Set(FAXDEST=/tmp/faxes)
3. Set(tempfax=${SHELL(ls /tmp/faxes/*.tif):11})
4. NoOp(file name is: ${tempfax})
5. Goto(incoming-fax,fax,7)
[ Context 'incoming-fax' created by 'pbx_config' ]
'fax' = 1. Verbose(3,Incoming fax)
...
5. ReceiveFax(${FAXDEST}/${tempfax})
6. Verbose(3,- Fax receipt completed with status:
${FAXSTATUS})
7. System(echo | /usr/bin/mutt -s New fax
earohua...@gmail.com -a ${FAXDEST}/${tempfax})
8. NoOp(System command status is: ${SYSTEMSTATUS})
9. Hangup()
ASTERISK CLI OUTPUT:
-- Goto (default,*95,1)
-- Executing [*95@default:1] NoOp(SIP/40106-1ea1, trying to send
a fax to an email) in new stack
-- Executing [*95@default:2] Set(SIP/40106-1ea1,
FAXDEST=/tmp/faxes) in new stack
-- Executing [*95@default:3] Set(SIP/40106-1ea1,
tempfax=20130619.tif) in new stack
-- Executing [*95@default:4] NoOp(SIP/40106-1ea1, file name is:
20130619.tif) in new stack
-- Executing [*95@default:5] Goto(SIP/40106-1ea1,
incoming-fax,fax,7) in new stack
-- Goto (incoming-fax,fax,7)
-- Executing [fax@incoming-fax:7] System(SIP/40106-1ea1, echo |
/usr/bin/mutt -s New fax earohua...@gmail.com -a
/tmp/faxes/20130619.tif) in new stack
-- Executing [fax@incoming-fax:8] NoOp(SIP/40106-1ea1, System
command status is: APPERROR) in new stack
-- Executing [fax@incoming-fax:9] Hangup(SIP/40106-1ea1, ) in
new stack
Elder D. Arohuanca
Lima - Peru
On Wed, Jun 19, 2013 at 1:38 PM, Andre Courchesne voipfor...@gmail.comwrote:
Why echo | ?
Alsy are you sire of the content of ${FAXDEST} and ${tempfax}.
Add some NoOp before.
On 2013-06-19, at 2:29 PM, Daniel - Asterisk earohua...@gmail.com wrote:
Hi Andre,
I've tried with:
System(echo | /usr/bin/mutt -s New fax earohua...@gmail.com -a
${FAXDEST}/${tempfax})
with no success, value of SYSTEMSTATUS variable is APPERROR
Again it works from Linux shell.
Thanks in advance
Elder
On Wed, Jun 19, 2013 at 1:08 PM, Andre Courchesne voipfor...@gmail.comwrote:
Probably Asterisk does not know where mutt is, specify it's path in your
System command.
On 2013-06-19, at 2:03 PM, Daniel - Asterisk earohua...@gmail.com
wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works, but when trying with Asterisk's System command it doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com -a
/tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com -a
${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by
root.
Any hint will be appreciated.
Elder D. Arohuanca
Lima - Peru
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Re: [asterisk-users] Mailing a fax with mutt does not succeed
More things to try:
(1) Is there any entry in /var/log/maillog (or equivalent log file)? If so, mutt basically works
and the messages should give some clues.
(2) What happens if you call mutt without any attachments?
I am using mutt in exactly the same way and it works.
jg
Am 19.06.2013 21:50, schrieb Daniel - Asterisk:
Hi Andre:
I added echo to provide STDIN, I'm sure on variable contents, please see bellow
Hello Steve,
1. I've just addd echo at my sentence, please see output bellow.
2. Asterisk is executing as root, I think Asterisk has access to read TIF files since I've
used ls, chmod, cp mv from Asterisk's CLI with '!' character.
3. I don't get you, please give some advice to try using Verbose instead System
4. I don't know how to get this, but I'm using /usr/bin/mutt as you can see
bellow.
5. I have redirected output of System this way : System(echo | /usr/bin/mutt -s New fax
earohua...@gmail.com mailto:earohua...@gmail.com -a ${FAXDEST}/${tempfax}
/tmp/ocurrencies.txt 21), ocurrencies.txt is empty.
DIALPLAN:
[ Context 'default' created by 'pbx_config' ]
'*95' = 1. NoOp(trying to send a fax to an email)
2. Set(FAXDEST=/tmp/faxes)
3. Set(tempfax=${SHELL(ls /tmp/faxes/*.tif):11})
4. NoOp(file name is: ${tempfax})
5. Goto(incoming-fax,fax,7)
[ Context 'incoming-fax' created by 'pbx_config' ]
'fax' = 1. Verbose(3,Incoming fax)
...
5. ReceiveFax(${FAXDEST}/${tempfax})
6. Verbose(3,- Fax receipt completed with status:
${FAXSTATUS})
7. System(echo | /usr/bin/mutt -s New fax earohua...@gmail.com
mailto:earohua...@gmail.com -a ${FAXDEST}/${tempfax})
8. NoOp(System command status is: ${SYSTEMSTATUS})
9. Hangup()
ASTERISK CLI OUTPUT:
-- Goto (default,*95,1)
-- Executing [*95@default:1] NoOp(SIP/40106-1ea1, trying to send a fax to an
email) in new stack
-- Executing [*95@default:2] Set(SIP/40106-1ea1,
FAXDEST=/tmp/faxes) in new stack
-- Executing [*95@default:3] Set(SIP/40106-1ea1, tempfax=20130619.tif) in new
stack
-- Executing [*95@default:4] NoOp(SIP/40106-1ea1, file name is: 20130619.tif)
in new stack
-- Executing [*95@default:5] Goto(SIP/40106-1ea1,
incoming-fax,fax,7) in new stack
-- Goto (incoming-fax,fax,7)
-- Executing [fax@incoming-fax:7] System(SIP/40106-1ea1, echo | /usr/bin/mutt -s
New fax earohua...@gmail.com mailto:earohua...@gmail.com -a /tmp/faxes/20130619.tif)
in new stack
-- Executing [fax@incoming-fax:8] NoOp(SIP/40106-1ea1, System command status is:
APPERROR) in new stack
-- Executing [fax@incoming-fax:9] Hangup(SIP/40106-1ea1, ) in new
stack
Elder D. Arohuanca
Lima - Peru
On Wed, Jun 19, 2013 at 1:38 PM, Andre Courchesne voipfor...@gmail.com
mailto:voipfor...@gmail.com wrote:
Why echo | ?
Alsy are you sire of the content of ${FAXDEST} and ${tempfax}.
Add some NoOp before.
On 2013-06-19, at 2:29 PM, Daniel - Asterisk earohua...@gmail.com
mailto:earohua...@gmail.com wrote:
Hi Andre,
I've tried with:
System(echo | /usr/bin/mutt -s New fax earohua...@gmail.com
mailto:earohua...@gmail.com -a ${FAXDEST}/${tempfax})
with no success, value of SYSTEMSTATUS variable is APPERROR
Again it works from Linux shell.
Thanks in advance
Elder
On Wed, Jun 19, 2013 at 1:08 PM, Andre Courchesne voipfor...@gmail.com
mailto:voipfor...@gmail.com wrote:
Probably Asterisk does not know where mutt is, specify it's path in
your System command.
On 2013-06-19, at 2:03 PM, Daniel - Asterisk earohua...@gmail.com
mailto:earohua...@gmail.com wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the
Linux shel it
works, but when trying with Asterisk's System command it doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com
mailto:earohua...@gmail.com -a
/tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com
mailto:elder.arohua...@gmail.com -a ${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by
root.
Any hint will be appreciated.
Elder D. Arohuanca
Lima - Peru
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Re: [asterisk-users] Mailing a fax with mutt does not succeed
This worked for me:
System(date | mutt -s FAX from
${CALLERID(num)} -a
/var/spool/asterisk/fax/${STRFTIME(,,%Y%m%d)}-${CALLERID(num)}.tiff
mail@domain)
I don't see any difference with you (beside the echo
instead of date), so I guess you should look at the maillog to find out
what is happening. Or (as I did in the end) write AGI script to send the
goddamn mail using the language and method you like (I used Perl and
MIME::Lite::TT::HTML), and make Asterisk call that script.
jg
писал 19.06.2013 23:12:
More things to try:
(1) Is there
any entry in /var/log/maillog (or equivalent log file)? If so, mutt
basically works and the messages should give some clues.
(2) What
happens if you call mutt without any attachments?
I am using mutt
in exactly the same way and it works.
jg
Am 19.06.2013 21:50,
schrieb Daniel - Asterisk:
Hi Andre:
I added echo to
provide STDIN, I'm sure on variable contents, please see bellow
Hello Steve,
1. I've just addd echo at my sentence, please see
output bellow.
2. Asterisk is executing as root, I think Asterisk
has access to read TIF files since I've used ls, chmod, cp mv from
Asterisk's CLI with '!' character.
3. I don't get you, please give
some advice to try using Verbose instead System
4. I don't know how
to get this, but I'm using /usr/bin/mutt as you can see bellow.
5. I
have redirected output of System this way : System(echo | /usr/bin/mutt
-s New fax earohua...@gmail.com -a ${FAXDEST}/${tempfax}
/tmp/ocurrencies.txt 21), ocurrencies.txt is empty.
DIALPLAN:
[ Context 'default' created by 'pbx_config' ]
'*95' = 1.
NoOp(trying to send a fax to an email)
2. Set(FAXDEST=/tmp/faxes)
3. Set(tempfax=${SHELL(ls /tmp/faxes/*.tif):11})
4. NoOp(file
name is: ${tempfax})
5. Goto(incoming-fax,fax,7)
[ Context
'incoming-fax' created by 'pbx_config' ]
'fax' = 1.
Verbose(3,Incoming fax)
...
5. ReceiveFax(${FAXDEST}/${tempfax})
6. Verbose(3,- Fax receipt completed with status: ${FAXSTATUS})
7. System(echo | /usr/bin/mutt -s New fax earohua...@gmail.com -a
${FAXDEST}/${tempfax})
8. NoOp(System command status is:
${SYSTEMSTATUS})
9. Hangup()
ASTERISK CLI OUTPUT:
-- Goto
(default,*95,1)
-- Executing [*95@default:1]
NoOp(SIP/40106-1ea1, trying to send a fax to an email) in new
stack
-- Executing [*95@default:2] Set(SIP/40106-1ea1,
FAXDEST=/tmp/faxes) in new stack
-- Executing [*95@default:3]
Set(SIP/40106-1ea1, tempfax=20130619.tif) in new stack
--
Executing [*95@default:4] NoOp(SIP/40106-1ea1, file name is:
20130619.tif) in new stack
-- Executing [*95@default:5]
Goto(SIP/40106-1ea1, incoming-fax,fax,7) in new stack
-- Goto
(incoming-fax,fax,7)
-- Executing [fax@incoming-fax:7]
System(SIP/40106-1ea1, echo | /usr/bin/mutt -s New fax
earohua...@gmail.com -a /tmp/faxes/20130619.tif) in new stack
--
Executing [fax@incoming-fax:8] NoOp(SIP/40106-1ea1, System
command status is: APPERROR) in new stack
-- Executing
[fax@incoming-fax:9] Hangup(SIP/40106-1ea1, ) in new stack
Elder D. Arohuanca
Lima - Peru
On Wed, Jun 19, 2013 at
1:38 PM, Andre Courchesne voipfor...@gmail.com wrote:
Why
echo | ?
Alsy are you sire of the content of ${FAXDEST} and
${tempfax}.
Add some NoOp before.
On 2013-06-19, at
2:29 PM, Daniel - Asterisk earohua...@gmail.com wrote:
Hi
Andre,
I've tried with:
System(echo | /usr/bin/mutt -s
New fax earohua...@gmail.com -a ${FAXDEST}/${tempfax})
with no success, value of SYSTEMSTATUS variable is APPERROR
Again it works from Linux shell.
Thanks in advance
Elder
On Wed, Jun 19, 2013 at 1:08 PM, Andre
Courchesne voipfor...@gmail.com wrote:
Probably Asterisk
does not know where mutt is, specify it's path in your System command.
On 2013-06-19, at 2:03 PM, Daniel - Asterisk
earohua...@gmail.com wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from
the Linux shel it works, but when trying with Asterisk's System command
it doesn't.
Successful Linux command:
echo |
mutt -s New fax earohua...@gmail.com -a /tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same =
n,System(mutt -s New fax elder.arohua...@gmail.com -a
${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0
on Debian 6.0.6, Asterisk was installed by root.
Any
hint will be appreciated.
Elder D. Arohuanca
Lima - Peru --
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Re: [asterisk-users] Mailing a fax with mutt does not succeed
Please don't top-post.
On Wed, 19 Jun 2013, Steve Edwards wrote:
1) Doesn't mutt expect the body on stdin? (Where's the 'echo' in the
Asterisk command?)
On Wed, 19 Jun 2013, Daniel - Asterisk wrote:
1. I've just addd echo at my sentence, please see output bellow.
Not that it's related to your issue...
Since mutt is expecting the body on stdin, why not put something in like
Hey, you've got FAX.
2) Is Asterisk executing as root? Does the Asterisk user ID have read
access to the TIFF?
2. Asterisk is executing as root, I think Asterisk has access to read
TIF files since I've used ls, chmod, cp mv from Asterisk's CLI with
'!' character.
This is not conclusive. The Asterisk 'daemon' is a separate process from
your CLI process (and your shell process spawned from the CLI process).
They do not 'have to' be executing as the same user. If the Asterisk
daemon is executing as root, permissions should not be an issue -- unless
you're doing something weird with /tmp/ and NFS.
3) If you use 'verbose()' instead of 'system()' does the command look
like your shell command?
3. I don't get you, please give some advice to try using Verbose instead
System
If you replace system() with verbose(), does the command you are trying to
execute look correct? If you 'copy' the command from the Asterisk CLI and
'paste' it into a shell and execute it using sudo as the the user
executing the Asterisk daemon, does it work?
4) Is mutt in the Asterisk user ID's path?
4. I don't know how to get this, but I'm using /usr/bin/mutt as you can
see bellow.
Then that should be sufficient.
5) If you redirect the output in the system() command to a file, does
that yield any clues? I.e., system(foo /tmp/clue 21)
5. I have redirected output of System this way : System(echo |
/usr/bin/mutt -s New fax earohua...@gmail.com -a ${FAXDEST}/${tempfax}
/tmp/ocurrencies.txt 21), ocurrencies.txt is empty.
Sometimes the 'littlest' thing can trip you up. Usually it comes down to
ownership, permissions, or environment variables.
Not that this is the proper way to figure this out, but this is what I
would try next in desperation -- after crossing my fingers.
(Cut and paste each line from this email to a shell.)
AUSER=$(ps --no-heading -C asterisk --format user)
FILE=/tmp/faxes/20130619.tif
MUSER=earohua...@gmail.com
echo body /tmp/body
env --ignore sudo -u ${AUSER} /usr/bin/file ${FILE}
env --ignore sudo -u ${AUSER} /usr/bin/mutt -v
env --ignore sudo -u ${AUSER} /usr/bin/mutt -s in-body ${MUSER} /tmp/body
env --ignore sudo -u ${AUSER} /usr/bin/mutt -a /tmp/body -s as-attachment-and-body
${MUSER} /tmp/body
env --ignore sudo -u ${AUSER} /usr/bin/mutt -a ${FILE} -s new-fax ${MUSER}
/tmp/body
Good luck :)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Asterisk / PHP-AGI / pthreads
On Mon, Jun 17, 2013 at 7:22 PM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 17 Jun 2013, Thorsten Göllner wrote:
does anyone have experience with Asterisk-AGI-Scripts in PHP while using
pthreads in PHP? Are there any limitations or problems known?
I've written 'pthread-ed' AGIs in C.
The only 'pthread related' limitation I stumbled into is that you can only
execute a single AGI request at a time -- which is kind of obvious if you
understand the AGI protocol.
My use case was playing a file ('Please wait while we authorize your
credit card') while processing the credit request.
Since our card processor almost always returned the credit response before
the end of the file, the 'user experience' was that the credit request was
instantaneous.
--
Thanks in advance,
--**--**
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
Steve,
Would you mind sharing a sample of your pthread-ed C AGI?
This will help someone like me who has written AGI in Perl/PHP and now
exploring C AGI.
Thanks,
--Satish
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