Re: [SR-Users] How to send "bye" to caller when callee abruptly terminates (Connection reset by peer)
Thank you it worked. From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Friday, December 22, 2017 2:24 AM To: Pranathi Venkatayogi ; Kamailio (SER) - Users Mailing List Subject: Re: How to send "bye" to caller when callee abruptly terminates (Connection reset by peer) Hello, you can use htable module to store the association between connection id (iirc, the var $conid) and dialog attributes (call id, from/to tag). The values in the hash table are available in the event route. Cheers, Daniel On 21.12.17 20:53, Pranathi Venkatayogi wrote: Hi, Currently when callee abruptly closes the connection (TCP RST), Kamailio does not send “BYE” to the caller. Caller is left to believe that he is talking to the callee. What is best way to solve this? I have attempted using tcpops module to track connection RSTs and then send $dlg_bye in event_route[tcp:reset] . In the event_route, I can access only these - $conid , $Ri , $Rp , $si , $sp and $proto variable and do not know how to get to the dlg this connection belongs. Questions: How do I load the dialog associated with this conid? Is there a way to enumerate all dialogs? Thanks in advance, Pranathi. -- Daniel-Constantin Mierla www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - www.asipto.com<http://www.asipto.com> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com<http://www.kamailioworld.com> ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] How to send "bye" to caller when callee abruptly terminates (Connection reset by peer)
Hi, Currently when callee abruptly closes the connection (TCP RST), Kamailio does not send “BYE” to the caller. Caller is left to believe that he is talking to the callee. What is best way to solve this? I have attempted using tcpops module to track connection RSTs and then send $dlg_bye in event_route[tcp:reset] . In the event_route, I can access only these - $conid , $Ri , $Rp , $si , $sp and $proto variable and do not know how to get to the dlg this connection belongs. Questions: How do I load the dialog associated with this conid? Is there a way to enumerate all dialogs? Thanks in advance, Pranathi. ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] re.subst does not work with $var?
Any input please? How do I use these methods with $var...
I get the same behavior when I use avp_subst as well.
From: Pranathi Venkatayogi
Sent: Thursday, June 15, 2017 6:22 PM
To: Kamailio (SER) - Users Mailing List
Subject: re.subst does not work with $var?
I am trying to replace "domain1" with "domain2" in the msrp(body).
Following stmt works where in I specify domain strings as literals.
var(modBody) = $(msrp(body){re.subst,/()(domain1)()/domain2/g});
I am having trouble making the same work with $var variables.
See below - this code does not do the replacement -
$var(domain1) = "()(domain1)()";
$var(domain2) = "domain2";
var(modBody2) = $(msrp(body){re.subst,/$var(domain1)/$var(domain2)/g});
Any clue/ideas or is it by design?
Thanks,
Pranathi Venkatayogi
System Developer II
(520) 745-9447 x4466
www.cyracom.com<http://www.cyracom.com/>
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Re: [SR-Users] How to change MSRP from sip address?
Yes. In the example below, I would like keep the same connection, but change
"From" when routing to the other end. And change it back when messages are
destined to cust3 again.
2017-06-15 18:12:22.622140 [blink.exe 18176]: 10.11.200.13:56459 -->
10.99.52.17:2855
MSRP 724e5edef489340c SEND
To-Path: msrps://10.99.52.17:2855/s.28641.1.1903524142;tcp
msrps://10.99.54.18:2855/e47976b4f6716ca7ce3a;tcp
From-Path: msrps://10.0.0.5:56459/bf85600695b5ab468931;tcp
Message-ID: 557e5ab42c43da38
Byte-Range: 1-667/667
Success-Report: yes
Failure-Report: yes
Content-Type: message/cpim
From: cust3
To:
DateTime: 2017-06-15T18:12:22.620140-05:00
MIME-Version: 1.0
Content-Type: text/html; charset="utf8"
http://www.w3.org/TR/REC-html40/strict.dtd";>
p, li { white-space: pre-wrap; }
new
one
---724e5edef489340c$
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Friday, June 16, 2017 1:50 AM
To: Pranathi Venkatayogi ; Kamailio (SER) - Users
Mailing List
Cc: Mick McGrellis
Subject: Re: How to change MSRP from sip address?
Hello,
what do you mean by 'from sip address'? The From header in the INVITE
initiating a MSRP chat?
Cheers,
Daniel
On 15.06.17 23:56, Pranathi Venkatayogi wrote:
I need to replace the "From" address of MSRP message with something different.
I noticed that module does not allow one to set values from config file.
What does it take to enable this functionality? Does anyone have a prototype I
can leverage?
Need to replace from domain with some other domain -
Given: MSRP body is - From: cust3
#015#012To:
#015#012DateTime:
.
Wanted: MSRP body is - From: cust3 #015#012To:
#015#012DateTime:
.
Thanks,
Pranathi Venkatayogi
System Developer II
(520) 745-9447 x4466
www.cyracom.com<http://www.cyracom.com/>
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[SR-Users] re.subst does not work with $var?
I am trying to replace "domain1" with "domain2" in the msrp(body).
Following stmt works where in I specify domain strings as literals.
var(modBody) = $(msrp(body){re.subst,/()(domain1)()/domain2/g});
I am having trouble making the same work with $var variables.
See below - this code does not do the replacement -
$var(domain1) = "()(domain1)()";
$var(domain2) = "domain2";
var(modBody2) = $(msrp(body){re.subst,/$var(domain1)/$var(domain2)/g});
Any clue/ideas or is it by design?
Thanks,
Pranathi Venkatayogi
System Developer II
(520) 745-9447 x4466
www.cyracom.com<http://www.cyracom.com/>
Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0]
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[SR-Users] How to change MSRP from sip address?
I need to replace the "From" address of MSRP message with something different. I noticed that module does not allow one to set values from config file. What does it take to enable this functionality? Does anyone have a prototype I can leverage? Need to replace from domain with some other domain - Given: MSRP body is - From: cust3 #015#012To: #015#012DateTime: . Wanted: MSRP body is - From: cust3 #015#012To: #015#012DateTime: . Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc. ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] How to gateway MSRP messages into XMPP
I read about XMPP module in Kamailio, but it looks like it works with "MESSAGE" packets (SIMPLE). How do I gateway MSRP packets to XMPP in Kamailio. Do we have module support? Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc. ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] dlg_refer caller fails to create callee dialog
Here is the kamcmd output. Strangely the state is showing as 4/terminated,
while I don’t see any bye message/logs to that extent in Kamailio. And my chat
session is still on.
kamcmd> dlg.list
{
h_entry: 3772
h_id: 5660
call-id: e6b6863121c7420cbded82574cff12d0my
from_uri: sip:cu...@devtranslation.sms-test.cyracom.com
to_uri: sip:span...@devtranslation.sms-test.cyracom.com
state: 4
start_ts: 1494343858
init_ts: 1494343857
timeout: 1494387058
lifetime: 43200
dflags: 643
sflags: 0
iflags: 0
caller: {
tag: 69786a5fef6a48bb89b8277cc485d82d
contact:
sip:71594386@10.11.200.59:49376;alias=10.11.200.59~49376~1
cseq: 9739
route_set:
socket: udp:10.99.52.17:5060
}
callee: {
tag: 408c77a154484b5083f5a895e8fc664b
contact: sip:48751293@10.99.54.28:55933
cseq: 0
route_set:
socket: udp:10.99.52.17:5060
}
profiles: {
}
variables: {
{
"agenturi":
sip:age...@devtranslation.sms-test.cyracom.com
}
}
}
kamcmd>
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
Daniel-Constantin Mierla
Sent: Monday, May 08, 2017 11:42 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] dlg_refer caller fails to create callee dialog
Hello,
can you provide the output of kamcmd dlg.list for the call just before doing
the refer?
Cheers,
Daniel
On 09.05.17 00:44, Pranathi Venkatayogi wrote:
Hi,
I am calling dlg_refer after dialog is confirmed on both sides. This time the
callee is a real client, so call was established end to end.
But still I get the error “No contact available” when callee dialog is being
created – see below.
I am attaching the snippet of my config where I am calling dlg_refer as well..
Is there something wrong with my usage? Please advise.
Thanks,
Pranathi
4719 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: INFO:
[SR-Users] dlg_refer caller fails to create callee dialog
Hi, I am calling dlg_refer after dialog is confirmed on both sides. This time the callee is a real client, so call was established end to end. But still I get the error "No contact available" when callee dialog is being created - see below. I am attaching the snippet of my config where I am calling dlg_refer as well.. Is there something wrong with my usage? Please advise. Thanks, Pranathi 4719 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: INFO:
Re: [SR-Users] How does dlg_bridge work?
I found the root cause of it. This is because I am terminating sessions on
Kamailio, by sending 200 ok from config. I don’t specify callee “Contact” at
that point. Even though there are active chat sessions, dialog are not shown
with cmd below.
I am attaching relevant config below -
Questions:
1. Can dlg_refer work without callee contact details? Or
2. Should I make up some contact and put in the 200 OK below Or
3. Should I handcraft a REFER message and send to caller instead?
Please advise.
Config section:
sl_send_reply("100", "Trying");
sl_send_reply("180", "Ringing");
append_to_reply("Record-Route: \r\n");
pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)");
$var(rbody) = "v=0\r\n";
$var(rbody) = $var(rbody) + "o=- 3688213899 3688213900 IN IP4
MY_IP_ADDR\r\n";
$var(rbody) = $var(rbody) + "s=Kamailio\r\n";
$var(rbody) = $var(rbody) + "t=0 0\r\n";
$var(rbody) = $var(rbody) + "m=message 2855 TCP/TLS/MSRP *\r\n";
$var(rbody) = $var(rbody) + "c=IN IP4 MY_IP_ADDR\r\n";
$var(rbody) = $var(rbody) + "a=path:msrps://MY_IP_ADDR:MY_MSRP_PORT/" +
$var(sessid) + ";tcp\r\n";
$var(rbody) = $var(rbody) + "a=accept-types:message/cpim text/* image/*
application/im-iscomposing+xml\r\n";
$var(rbody) = $var(rbody) + "a=accept-wrapped-types:text/* image/*
application/im-iscomposing+xml\r\n";
$var(rbody) = $var(rbody) + "a=setup:passive\r\n";
set_reply_body("$var(rbody)", "application/sdp");
sl_send_reply("200", "OK");
Thanks
Pranathi
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Friday, May 05, 2017 2:43 AM
To: Kamailio (SER) - Users Mailing List ; Pranathi
Venkatayogi
Subject: Re: [SR-Users] How does dlg_bridge work?
Hello,
is the dialog already answered at that time? Can you see it via kamcmd dlg.list
?
Cheers,
Daniel
On 04.05.17 22:48, Pranathi Venkatayogi wrote:
When I use dlg_refer I get the following error:
2689 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: exec: ***
cfgtrace:failure_route=[HTTP_ALLOCATE_RESPONSE] c=[routingdev.cfg] l=567 a=26
n=dlg_refer
2690 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog
[dlg_hash.c:690]: dlg_lookup(): ref dlg 0x7f92d19835e8 with 1 -> 2
2691 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog
[dlg_hash.c:692]: dlg_lookup(): dialog id=10603 found on entry 2058
2692 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog
[dlg_hash.c:452]: build_new_dlg(): new dialog on hash 2058
2693 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog
[dlg_req_within.c:85]: build_dlg_t(): no contact available
2694 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog
[dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t
2695 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog
[dlg_hash.c:922]: dlg_unref_helper(): unref op on 0x7f92d19835e8 with 1 from
dlg_hash.c:940
From: Pranathi Venkatayogi
Sent: Thursday, May 04, 2017 5:28 AM
To: 'Kamailio (SER) - Users Mailing List'
<mailto:sr-users@lists.kamailio.org>
Subject: How does dlg_bridge work?
I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is
logged in, but is not in any call yet.
I want now refer cust1@somedomain to talk to agent directly. How do I do that?
I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)","");
I get an “audio call” from “Kamailio” to customer, but nothing happens on the
agent side.
I tried “dlg_refer” as well but nothing happens at all.
Should I handcraft “Refer” message using uac module and send to “customer”
instead?
Please suggest.
Thanks
Pranathi Venkatayogi
System Developer II
(520) 745-9447 x4466
www.cyracom.com<http://www.cyracom.com/>
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Re: [SR-Users] How does dlg_bridge work?
When I use dlg_refer I get the following error:
2689 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: exec: ***
cfgtrace:failure_route=[HTTP_ALLOCATE_RESPONSE] c=[routingdev.cfg] l=567 a=26
n=dlg_refer
2690 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog
[dlg_hash.c:690]: dlg_lookup(): ref dlg 0x7f92d19835e8 with 1 -> 2
2691 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog
[dlg_hash.c:692]: dlg_lookup(): dialog id=10603 found on entry 2058
2692 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog
[dlg_hash.c:452]: build_new_dlg(): new dialog on hash 2058
2693 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog
[dlg_req_within.c:85]: build_dlg_t(): no contact available
2694 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog
[dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t
2695 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog
[dlg_hash.c:922]: dlg_unref_helper(): unref op on 0x7f92d19835e8 with 1 from
dlg_hash.c:940
From: Pranathi Venkatayogi
Sent: Thursday, May 04, 2017 5:28 AM
To: 'Kamailio (SER) - Users Mailing List'
Subject: How does dlg_bridge work?
I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is
logged in, but is not in any call yet.
I want now refer cust1@somedomain to talk to agent directly. How do I do that?
I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)","");
I get an "audio call" from "Kamailio" to customer, but nothing happens on the
agent side.
I tried "dlg_refer" as well but nothing happens at all.
Should I handcraft "Refer" message using uac module and send to "customer"
instead?
Please suggest.
Thanks
Pranathi Venkatayogi
System Developer II
(520) 745-9447 x4466
www.cyracom.com<http://www.cyracom.com/>
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[SR-Users] How does dlg_bridge work?
I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is
logged in, but is not in any call yet.
I want now refer cust1@somedomain to talk to agent directly. How do I do that?
I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)","");
I get an "audio call" from "Kamailio" to customer, but nothing happens on the
agent side.
I tried "dlg_refer" as well but nothing happens at all.
Should I handcraft "Refer" message using uac module and send to "customer"
instead?
Please suggest.
Thanks
Pranathi Venkatayogi
System Developer II
(520) 745-9447 x4466
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[SR-Users] How do we send "MSRP Message" from config
While implementing Kamailio to hold customer call, I would to send "Pls hold on looking for an agent" MSRP message from Kamailio config. Is it doable? >From the module, we can only send msrp_reply and msrp_relay... Please advise. Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc. ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Query with regard to http_client and http_async_client
I recently successfully used http_async_client for POST. Will try to help you
out here -
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
Ramachandran, Agalya (Contractor)
Sent: Tuesday, April 25, 2017 10:17 AM
To: giacomo.va...@gmail.com; sr-users@lists.kamailio.org
Subject: [SR-Users] Query with regard to http_client and http_async_client
Hi Giacomo,
I have few questions with regard to rest client in kamailio.
1) I see there are two modules in kamailio for rest client. http_client,
http_async_client.
I hope http_client does the http transfer in synchronous way.
http_async_client does the http transfer in asynchronous manner.
Correct me if am wrong here.
Correct.
2) I want to achieve https transfers in async way, so choosing
http_async_client.
Is there a way that kamailio maintains persistent TCP connection pooling for
http transfer.
I see there is a "workers" parameter in http_async_client. By increasing this
number does it result it more http request's handling?
Let the author reply here.
3) Also, I have some questions regarding documentation for module
http_async_client in 5.0 version.
In the API exposure, it says "http_async_query(url, route_name)".
But the example if I want to use POST, it says
"http_async_query("http://example.com/test.php";, "{'r_uri':'$rU',
'f_uri':'$fU'}", "HTTP_REPLY");"
There is a parameter mismatch between the example and the actual function.
If I try to use GET query, kamailio service is starting. But if I try to give
the POST query with payload, kamailio service is not started.
It says, sctp_core_destroy(): SCTP API not initialized.
Also I have noticed that, functions like http_append_header(),http_set_method()
are removed in 5.0 version, whereas we have this API's is 4.x versions.
Is there some issue with the documentation for 5.0 ?
I used it like this and it works -
http_async_query("http://TEXT_BROKER/api/events/CallAnswered";,
"{'CallId':'$ci', 'Time' : '$TF'}",
"HTTP_REPLY");
Issue may be with quotes. Make sure you don't have " in the second parameter
content. If you send Kamailio log it will be easier to debug.
Please let me the know the details for the above queries.
Regards,
Agalya
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Re: [SR-Users] How to eliminate Content-Length:0 from sl_send_reply
Hi Daniel, Just wanted to confirm my understanding here. Are you hinting towards INVITE (hold) message illustrated here (http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-01.html) Here is how it lays out: 1. Customer sends INVITE requiring MSRP chat 2. Kamailio will accept the invite and complete the session setup - send 100 Trying, 180 Ringing and 200 ok. 3. Kamailio will send INVITE(hold) to customer 4. When the agent becomes available, Kamailio will send "refer" to connect to right agent directly. Please share your comments on this. Thanks Pranathi. -Original Message- From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Daniel-Constantin Mierla Sent: Tuesday, April 25, 2017 9:09 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] How to eliminate Content-Length:0 from sl_send_reply Actually the scenario might work with some bits of coding, in the idea that the INVITE is answered for the purpose of keeping it on hold, so no real communication happens. Moreover, if I got all the details properly, here is about MSRP (text messaging), not audio/video session. Cheers, Daniel On 25.04.17 15:47, Alex Balashov wrote: > It's a legitimate aspiration, just not one that Kamailio is designed to > service, since it's a proxy - it just passes messages. For instance, one > cannot "terminate" an invite to Kamailio "for a while". As a messenger, it > has no means of doing that. You need an actual user agent involved. > > -- Alex > > -- > Principal, Evariste Systems LLC (www.evaristesys.com) > > Sent from my Google Nexus. > > ___ > Kamailio (SER) - Users Mailing List > sr-users@lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to eliminate Content-Length:0 from sl_send_reply
I want to place the customer call on hold while I look for the "agent" to
answer the chat call.
Daniel confirms that for a brief moment I can hold the calls on Kamailio.
Eventually we are looking for "FreeSwitch" to help us.
Hence this is what I am trying:
1) When customer invite comes - let it terminate on Kamailio for a while
- If msrp message comes from customer, reply "pls hold for agent to be
available" from Kamailio
2) When the agent becomes available, send "Refer" to customer to connect back
again.
- Now customer and agent can talk directly to each other.
So, for the step 1, I am generating SDP from Kamailio so Customer has
everything setup properly.
How do I do that from config?
If this is not right way to place a MSRP chat call on hold, what is the right
way. Please suggest.
Thanks
Pranathi
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex
Balashov
Sent: Tuesday, April 25, 2017 8:31 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] How to eliminate Content-Length:0 from sl_send_reply
Why are you generating your own SDP body? What are you trying to accomplish?
Every once in a while there is a legitimate reason, but 99% of the time that is
the wrong approach to whatever problem you're trying to solve.
-- Alex
> On Apr 25, 2017, at 9:06 AM, Pranathi Venkatayogi
> wrote:
>
> Hi Alex,
> I am adding my own SDP body to the reply and set the content-length to 350
> already.
> The extra content-length: 0 is causing it to be malformed.
>
> Question: Is there a way to generate SIP reply with SDP payload from config
> file. I saw there is uac_send_req but no support for reply so far...Please
> suggest.
>
> I want to send this message from Kamailio:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.0.0.5:52653;rport=52653;branch=z9hG4bKPj60f74c5d68574dc5ae3cbfdd21b
> a5f12;received=10.11.200.48
> From: "cust1"
> ;tag=0cba06eab62d47d1ac
> e9c8834bbe0396
> To:
> ;tag=02eb90a2953b2668
> 200237ca6e13-e339
> Call-ID: 7d383971684b44f792cfe84043bf8b5a
> CSeq: 15351 INVITE
> Content-Type: application/sdp
> Content-Length: 350
>
> v=0
> o=- 3688213899 3688213900 IN IP4 10.99.52.17 s=Blink 1.4.2 (Windows)
> t=0 0
> m=message 2855 TCP/MSRP *
> c=IN IP4 10.99.52.17
> a=path:msrps://10.99.52.17:2855/12673656159856;tcp
> a=accept-types:message/cpim text/* image/*
> application/im-iscomposing+xml
> a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml
> a=setup:active
>
> -Original Message-
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf
> Of Alex Balashov
> Sent: Tuesday, April 25, 2017 7:24 AM
> To: Kamailio (SER) - Users Mailing List
> Subject: Re: [SR-Users] How to eliminate Content-Length:0 from
> sl_send_reply
>
> Hello,
>
> What leads you to the conclusion that the
>
> Content-Length: 0
>
> header is "extra", and in need of a "workaround"?
>
> -- Alex
>
>> On Tue, Apr 25, 2017 at 12:21:32PM +, Pranathi Venkatayogi wrote:
>>
>> Hi,
>> How do I eliminate Content-Length:0 from sip reply when using sl_send_reply?
>> I wanted to generate 200 ok, with SDP payload. I appended headers like
>> below. The packet seems ok except there is extra "Content-Length:0", which
>> is causing problems. Any workarounds?
>>
>> The packet being generated is: Note the extra content-length: 0 at the end.
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP
>> 10.0.0.12:59404;rport=59404;branch=z9hG4bKPjb7d0391f8f7f479b812f6e5fe
>> a
>> dfa3c5;received=10.11.200.46
>> From: "cust1"
>> ;tag=afcffa277e2a43dc8
>> 8
>> fafc612f538864
>> To:
>> ;tag=37b119f34ca40c1
>> 5
>> 8c5b7306e1aa8d72.b73a
>> Call-ID: 7f73e5d03e15425395fac930eae423bd
>> CSeq: 31419 INVITE
>> Content-Type: application/sdp
>> Content-Length: 350
>>
>> v=0
>> o=- 3688213899 3688213900 IN IP4 10.99.52.17 s=Blink 1.4.2 (Windows)
>> t=0 0
>> m=message 2855 TCP/MSRP *
>> c=IN IP4 10.99.52.17
>> a=path:msrps://10.99.52.17:2855/116701522521345;tcp
>> a=accept-types:message/cpim text/* image/*
>> application/im-iscomposing+xml
>> a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml
>> a=setup:active
>> Content-Length: 0
>>
>> Code in config is:
>>append_to_reply("Content-Type: application/sdp\r\n");
>>append_to_reply("Content-Length: 350\r\n");
>>append_to_reply("\r\n");
>>append_to_reply(&qu
Re: [SR-Users] How to eliminate Content-Length:0 from sl_send_reply
Hi Alex,
I am adding my own SDP body to the reply and set the content-length to 350
already.
The extra content-length: 0 is causing it to be malformed.
Question: Is there a way to generate SIP reply with SDP payload from config
file. I saw there is uac_send_req but no support for reply so far...Please
suggest.
I want to send this message from Kamailio:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.0.0.5:52653;rport=52653;branch=z9hG4bKPj60f74c5d68574dc5ae3cbfdd21ba5f12;received=10.11.200.48
From: "cust1"
;tag=0cba06eab62d47d1ace9c8834bbe0396
To:
;tag=02eb90a2953b2668200237ca6e13-e339
Call-ID: 7d383971684b44f792cfe84043bf8b5a
CSeq: 15351 INVITE
Content-Type: application/sdp
Content-Length: 350
v=0
o=- 3688213899 3688213900 IN IP4 10.99.52.17
s=Blink 1.4.2 (Windows)
t=0 0
m=message 2855 TCP/MSRP *
c=IN IP4 10.99.52.17
a=path:msrps://10.99.52.17:2855/12673656159856;tcp
a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml
a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml
a=setup:active
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex
Balashov
Sent: Tuesday, April 25, 2017 7:24 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] How to eliminate Content-Length:0 from sl_send_reply
Hello,
What leads you to the conclusion that the
Content-Length: 0
header is "extra", and in need of a "workaround"?
-- Alex
On Tue, Apr 25, 2017 at 12:21:32PM +, Pranathi Venkatayogi wrote:
> Hi,
> How do I eliminate Content-Length:0 from sip reply when using sl_send_reply?
> I wanted to generate 200 ok, with SDP payload. I appended headers like
> below. The packet seems ok except there is extra "Content-Length:0", which is
> causing problems. Any workarounds?
>
>The packet being generated is: Note the extra content-length: 0 at the end.
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.0.0.12:59404;rport=59404;branch=z9hG4bKPjb7d0391f8f7f479b812f6e5fea
> dfa3c5;received=10.11.200.46
> From: "cust1"
> ;tag=afcffa277e2a43dc88
> fafc612f538864
> To:
> ;tag=37b119f34ca40c15
> 8c5b7306e1aa8d72.b73a
> Call-ID: 7f73e5d03e15425395fac930eae423bd
> CSeq: 31419 INVITE
> Content-Type: application/sdp
> Content-Length: 350
>
> v=0
> o=- 3688213899 3688213900 IN IP4 10.99.52.17 s=Blink 1.4.2 (Windows)
> t=0 0
> m=message 2855 TCP/MSRP *
> c=IN IP4 10.99.52.17
> a=path:msrps://10.99.52.17:2855/116701522521345;tcp
> a=accept-types:message/cpim text/* image/*
> application/im-iscomposing+xml
> a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml
> a=setup:active
> Content-Length: 0
>
> Code in config is:
> append_to_reply("Content-Type: application/sdp\r\n");
> append_to_reply("Content-Length: 350\r\n");
> append_to_reply("\r\n");
> append_to_reply("v=0\r\n");
> append_to_reply("o=- 3688213899 3688213900 IN IP4 MY_IP_ADDR\r\n");
> append_to_reply("s=Blink 1.4.2 (Windows)\r\n");
> append_to_reply("t=0 0\r\n");
> append_to_reply("m=message 2855 TCP/MSRP *\r\n");
> append_to_reply("c=IN IP4 MY_IP_ADDR\r\n");
>
> append_to_reply("a=path:msrps://MY_IP_ADDR:2855/$var(ssessid);tcp\r\n");
> append_to_reply("a=accept-types:message/cpim text/* image/*
> application/im-iscomposing+xml\r\n");
> append_to_reply("a=accept-wrapped-types:text/* image/*
> application/im-iscomposing+xml\r\n");
> append_to_reply("a=setup:active\r\n");
>
> sl_send_reply("200", "OK");
>
> Thanks,
> Pranathi Venkatayogi
> System Developer II
> (520) 745-9447 x4466
> www.cyracom.com<http://www.cyracom.com/>
>
> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0]
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>
> [cid:image005.png@01D1C9C9.C4E8B6D0]
>
> DISCLAIMER: This e-mail and any attached content may contain confidential or
> privileged material delivered for the sole use of the intended recipient(s).
> Any review, use, distribution or disclosure by others is strictly prohibited.
> If you are not the intended recipient (or authorized to receive for the
> recipient), please contact the sender immediately by reply e-mail and delete
> all copies of this message. It is the recipient's responsibility to scan
[SR-Users] How to eliminate Content-Length:0 from sl_send_reply
Hi,
How do I eliminate Content-Length:0 from sip reply when using sl_send_reply?
I wanted to generate 200 ok, with SDP payload. I appended headers like below.
The packet seems ok except there is extra "Content-Length:0", which is causing
problems. Any workarounds?
The packet being generated is: Note the extra content-length: 0 at the end.
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.0.0.12:59404;rport=59404;branch=z9hG4bKPjb7d0391f8f7f479b812f6e5feadfa3c5;received=10.11.200.46
From: "cust1"
;tag=afcffa277e2a43dc88fafc612f538864
To:
;tag=37b119f34ca40c158c5b7306e1aa8d72.b73a
Call-ID: 7f73e5d03e15425395fac930eae423bd
CSeq: 31419 INVITE
Content-Type: application/sdp
Content-Length: 350
v=0
o=- 3688213899 3688213900 IN IP4 10.99.52.17
s=Blink 1.4.2 (Windows)
t=0 0
m=message 2855 TCP/MSRP *
c=IN IP4 10.99.52.17
a=path:msrps://10.99.52.17:2855/116701522521345;tcp
a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml
a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml
a=setup:active
Content-Length: 0
Code in config is:
append_to_reply("Content-Type: application/sdp\r\n");
append_to_reply("Content-Length: 350\r\n");
append_to_reply("\r\n");
append_to_reply("v=0\r\n");
append_to_reply("o=- 3688213899 3688213900 IN IP4 MY_IP_ADDR\r\n");
append_to_reply("s=Blink 1.4.2 (Windows)\r\n");
append_to_reply("t=0 0\r\n");
append_to_reply("m=message 2855 TCP/MSRP *\r\n");
append_to_reply("c=IN IP4 MY_IP_ADDR\r\n");
append_to_reply("a=path:msrps://MY_IP_ADDR:2855/$var(ssessid);tcp\r\n");
append_to_reply("a=accept-types:message/cpim text/* image/*
application/im-iscomposing+xml\r\n");
append_to_reply("a=accept-wrapped-types:text/* image/*
application/im-iscomposing+xml\r\n");
append_to_reply("a=setup:active\r\n");
sl_send_reply("200", "OK");
Thanks,
Pranathi Venkatayogi
System Developer II
(520) 745-9447 x4466
www.cyracom.com<http://www.cyracom.com/>
Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0]
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Any review, use, distribution or disclosure by others is strictly prohibited.
If you are not the intended recipient (or authorized to receive for the
recipient), please contact the sender immediately by reply e-mail and delete
all copies of this message. It is the recipient's responsibility to scan this
e-mail and any attachments for viruses. The content of this e-mail message,
including any attachments, does not comprise a contract or a portion of a
contract, and so does not bind CyraCom International, Inc. or any of its agents
or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly
owned subsidiaries of CyraCom International, Inc.
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[SR-Users] http_async_query in event_route[dialog:start] does not work
Hi,
I wanted to make an http_async_query in event_route[dialog:start] after the
call the confirmed. I see that flow does not even get to making
"async_send_query". Perhaps it is failing at the first line "if (msg==null)" in
w_http_async_post as I don't see any other error messages from the module.
I can confirm that my setup and api server are fine as other calls I make
from request_route
and event_route[dialog:end] are working.
I am attaching herewith my config files and Kamailio.log file. Relevant prob
spot is below-
Apr 20 17:37:17 devsip00 /sbin/kamailio[6286]: {2 3094 INVITE
d9695149ce484811ab98ca41a4e8c694} INFO:
Re: [SR-Users] How do i build MSRP chat bot?
Hi Daniel, We are looking for just MSRP. We already have a different solution for phone calls. Yes similar to call queuing system. Do you have any pointers? Thanks Pranathi From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Thursday, April 13, 2017 1:36 AM To: m...@dopensource.com Cc: Pranathi Venkatayogi ; Kamailio (SER) - Users Mailing List Subject: Re: How do i build MSRP chat bot? Hello, On 10.04.17 11:53, m...@dopensource.com<mailto:m...@dopensource.com> wrote: Hey Daniel, She wants to put them on hold and ideally send them a MSRP message saying “Please wait for us to find a person to speak with”. is it for calls that involve also voice communications, or just MSRP? Because it looks like you look for a call queuing system. Cheers, Daniel Once someone becomes free, we want to bridge the session. We found that if the MSRP Client doesn’t connect within a certain period of time it kills the session like any SIP INVITE. One approach is to create a MSRP bot that answers the call and then forwards the session once a person becomes free. We are looking for other approaches. Any thoughts? On Apr 10, 2017, at 5:36 AM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, the standard MSRP session starts with and INVITE. What do you want to achieve, answer the invite, have a chat and then reconnect the caller to another MSRP endpoint? Or hold a bit the INVITE until the target connects to the system (e.g., after a push notification)? Cheers, Daniel On 07.04.17 23:09, Pranathi Venkatayogi wrote: I would like to place the customer connecting via SIP/MSRP on hold until I find the right destination. What is the best way to do this? One idea is to write a MSRP bot. Does anyone know how to go about this? Other smart ideas? Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> <https://twitter.com/cyracom> <http://www.linkedin.com/company/cyracom> <http://www.cyracom.com/blog/> DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc. -- Daniel-Constantin Mierla www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com<http://www.asipto.com/> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com<http://www.kamailioworld.com/> -- Daniel-Constantin Mierla www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com<http://www.asipto.com> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com<http://www.kamailioworld.com> ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
