Thanks for your help Sammy, the call is working, and I will share later about
the change that fix this problem
From: sr-users on behalf of malik
sherif
Sent: Friday, February 12, 2016 9:13 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Fw
Hello Sammy,
Do you have my kamailio cfg file?
Thanks
Abdul
From: sr-users on behalf of malik
sherif
Sent: Friday, February 12, 2016 9:08 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC
,
Sammy.
On Fri, Feb 12, 2016 at 2:33 PM, malik sherif
mailto:asheri...@hotmail.com>> wrote:
Thanks Sammy, I will use pastebin.com<http://pastebin.com> next as you
recommended.
Thanks
Abdul
From: sr-users
mailto:sr-users-boun...@lists.sip-rout
te tool since 2002. Pastebin is a website
where you can store text online for a set period of time.
On Feb 11, 2016 18:32, "malik sherif"
mailto:asheri...@hotmail.com>> wrote:
While the full debug log is being approved, I just copy and paste some of the
log.
ready modified the packet but I still
need to take a look at the sip captures to verify this.
Thanks,
Sammy
On Thu, Feb 11, 2016 at 12:28 PM, malik sherif
mailto:asheri...@hotmail.com>> wrote:
Hello Sammy,
I used both the gateway method and external, the result is the same it goes th
domain part in RURI in one of my previous emails
looking at your sip traces, you've already modified the packet but I still
need to take a look at the sip captures to verify this.
Thanks,
Sammy
On Thu, Feb 11, 2016 at 12:28 PM, malik sherif
mailto:asheri...@hotmail.com>> wrote:
Hel
27;ve a feeling that this email should be in Freeswitch mailing list, not in
Kamailio's/
Regards,
Sammy
On Wed, Feb 10, 2016 at 5:00 PM, malik sherif
mailto:asheri...@hotmail.com>> wrote:
Hello,
I am using Kamailio and freeswitch to setup SBC but the I attempted to make a
call
ur FS the
domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server.
I've a feeling that this email should be in Freeswitch mailing list, not in
Kamailio's/
Regards,
Sammy
On Wed, Feb 10, 2016 at 5:00 PM, malik sherif
mailto:asheri...@hotmail.com>> wrot
Regards,
Sammy
On Fri, Jan 29, 2016 at 11:47 AM, SamyGo
mailto:govoi...@gmail.com>> wrote:
Hi Abdul,
This is where you are getting your 404 NOT Found from Kamailio:
On Thu, Jan 28, 2016 at 4:30 PM, malik sherif
mailto:asheri...@hotmail.com>> wrote:
I will also run the comm
a global siptrace on
Once you execute the above command make a call to destination and see what
FreeeSWITCH is trying to do.
Thanks,
Sammy.
On Thu, Jan 28, 2016 at 11:23 AM, malik sherif
mailto:asheri...@hotmail.com>> wrote:
Any hint?
From: sr-users
ma
you enable the sofia sip debug by using the following command.
fs_cli> sofia global siptrace on
Once you execute the above command make a call to destination and see what
FreeeSWITCH is trying to do.
Thanks,
Sammy.
On Thu, Jan 28, 2016 at 11:23 AM, malik sherif
m
Any hint?
From: sr-users on behalf of malik
sherif
Sent: Tuesday, January 26, 2016 11:35 PM
To: Kamailio (SER) - Users Mailing List; mico...@gmail.com
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Thanks again and here is the pcap file
Thanks again and here is the pcap file.
Thanks
Abdul
From: Daniel-Constantin Mierla
Sent: Friday, January 22, 2016 8:46 AM
To: malik sherif; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Can you attach
From: sr-users on behalf of malik
sherif
Sent: Wednesday, January 20, 2016 9:55 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2
689991
;tag=1378608-a163217-13c4-17e5-662d0efc-17e5
To: ;tag=vXSQDHXU3v36B
Call-ID: 137c190-a163217-13c4-17e5-3d0913f8-1...@abdulkamailiosip.com
CSeq: 2 ACK
Via: SIP/2.0/UDP
10.22.52.2;branch=z9hG4bK2722.5b81c7f3e4a61ce57604a60cda694802.0
Via: SIP/2.0/UDP 10.22.50.23:5060;branch=z9hG4bK-17e5-5d580e-42
9/01/16 23:44, malik sherif wrote:
Any idea as to how to correct this problem? Also wireshark trace shows that
unrecognized SIP header for invite from freeswitch to kamailio and the header
from freeswitch is X-FS-support: update _display ,send_info. How can I disable
this header? any othe
-users on behalf of malik
sherif
Sent: Wednesday, January 13, 2016 9:21 PM
To: Kamailio (SER) - Users Mailing List; mico...@gmail.com
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Kamailio SIP is sending 404 not found since freeswitch is generated
UN-allocated number, the
Kamailio SIP is sending 404 not found since freeswitch is generated
UN-allocated number, the call got rejected and goes to voicemail.
Thank you again for your help
Abdul
From: Daniel-Constantin Mierla
Sent: Wednesday, January 13, 2016 9:06 PM
To: malik sherif
).
From: sr-users on behalf of malik
sherif
Sent: Wednesday, January 13, 2016 8:11 PM
To: Kamailio (SER) - Users Mailing List; mico...@gmail.com
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
is there a new to edit vars.xml file? I haven't touched
s on behalf of malik
sherif
Sent: Wednesday, January 13, 2016 5:15 PM
To: Kamailio (SER) - Users Mailing List; mico...@gmail.com
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Thanks again Daniel for replying.
Now the call is failing with 404 not found and goes to voicemail.
My extensions are as follow:
##
From: Daniel-Constantin Mierla
Sent: Wednesday, January 13, 2016 6:34 AM
To: malik sherif; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Us
o and freeswitch integration for SBC
Hello,
did you configure the freeswitch to listen on loopback? You would need to do
bridging of singnaling and eventually rtp between the network interface and
loopback if you want this kind of topology.
Cheers,
Daniel
On 12/01/16 19:00, malik sherif wrote:
Hello
Any body has kamailio/freeswitch SBC working?
From: sr-users on behalf of malik
sherif
Sent: Tuesday, January 12, 2016 6:00 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Hello Abdul Basit,
I
behalf of malik
sherif
Sent: Monday, January 11, 2016 5:03 PM
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Hello,
Any hint? How do I get a response? is it through user digest?
Thanks
Abdulmalik Sherif
From
/freeswitch
From: sr-users on behalf of malik
sherif
Sent: Monday, January 11, 2016 5:03 PM
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC
Hello,
Any hint? How do I get a response? is it through user
Hello,
Any hint? How do I get a response? is it through user digest?
Thanks
Abdulmalik Sherif
From: sr-users on behalf of malik
sherif
Sent: Friday, January 8, 2016 7:39 PM
To: sr-users@lists.sip-router.org
Subject: [SR-Users] Kamailio and freeswitch
; If you want to actually override the From display value, that's a bit
> > more complex, since proxies aren't technically supposed to do that.
> > However, the 'uac' module gives you this capability:
> >
> >
> > http://kamailio.org/docs/modules/4.1.x/m
odule gives you this capability:
> >
> >
> > http://kamailio.org/docs/modules/4.1.x/modules/uac.html#uac.f.uac_replace_from
> >
> >
> > e.g.
> >
> > uac_replace_from("\"SHERIF MALIK\"", ""); # Don't modify From UR
Hello,
I set SIP users using kamctl add command with username ( I put the phone
numbers) doamin (doamin name) and password. Is their a way to add caller id
name? when I make a call , i see the phone number but for caller id name I
displayline1.
Thank you for your help.
Abdul
http://kb.asipto.com/siremis:install40x:main#step_2
There are 4 options there, you have to select all of them first
time.
You should drop siremis database if you created some tables
manually.
Cheers,
Daniel
On 17/03/14 19:54
Thank Hugo,
I did that but didn't work. one thing am going to check before I try, I will
need to check the privilege grant status of siremis and then hopefully it will
work.
Thanks
Abdul
From: hugo.servel...@gmail.com
To: sr-users@lists.sip-router.org
Date: Mon, 17 Mar 2014 14:48:14 -0600
Subjec
From: asheri...@hotmail.com
To: sr-users@lists.sip-router.org
Date: Mon, 17 Mar 2014 16:28:46 +
Subject: [SR-Users] login to siremis failed 'siremis.user' doesn't exist
When I attempted to login to siremis , i got the following errors "Base table
or view not found: 1146 Table 'siremis.u
When I attempted to login to siremis , i got the following errors "Base table
or view not found: 1146 Table 'siremis.user' doesn't exist"
And then I created the table manually using the following commands
CREATE TABLE `user` (
`SYSID` int(11) NOT NULL auto_increment,
`USERID` varchar(15
exit;
}
?>
If the file install.lock not exists ... redirect to directory
install ;-)
If it is not there ... place an empty file install.lock in the
siremis directory manually.
Regards
Raine
n the
siremis directory manually.
Regards
Rainer
Am 14.03.2014 05:45, schrieb malik sherif:
any body experienced problem having login window
popup?
From: asheri...@hotmail.com
To: sr-users@lists.sip-r
any body experienced problem having login window popup?
From: asheri...@hotmail.com
To: sr-users@lists.sip-router.org
Date: Thu, 13 Mar 2014 22:36:22 +
Subject: Re: [SR-Users] SIREMIS redirecting to login prompt issue
Did anybody experienced this problem?
From: asheri...@hotmail.com
To: s
Did anybody experienced this problem?
From: asheri...@hotmail.com
To: sr-users@lists.sip-router.org
Subject: RE: [SR-Users] SIREMIS redirecting to login prompt issue
Date: Tue, 11 Mar 2014 15:49:06 +
step 1:
Step 2
Step 3
STEP 4
Last step:
After this login widow should have redirect
looked this [1] already?
I would recommend dropping trixbox is favor of a plain Asterisk installation.
It makes things easier to configure.
[1] http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
Regards,
On Wed, Mar 12, 2014 at 9:47 AM, malik sherif wrote:
I am
I am looking documentation as to how to integrate kamailio and Asterisk, I am
not sure if I use the correct term "media server" but I would like to configure
3-way call, call waiting, call transfer , and call forwarding.
Your help is greatly appreciated.
Thanks
Abdul
/devel/modules/xhttp_rpc.html
It's a simple built in web interface for all rpc commands.
Regards,Ovidiu Sas
On Tue, Mar 11, 2014 at 11:02 AM, malik sherif wrote:
Any recommendation web interface for gathering statistics other than SIREMIS? I
installed SIREMIS but unable to resolve
step 1:
Step 2
Step 3
STEP 4
Last step:
After this login widow should have redirected to login widow for example
installation documentation is available on
http://kb.asipto.com/siremis:install40x:main
Thanks
Abdul
> Date: Tue, 11 Mar 2014 11:40:09 -0400
> From: abal
After it display this window
after 10 second it returning me to steps 1 or this window also if I click on
launch Siremis it goes back to step1
Thanks again.
Abdul
> Date: Tue, 11 Mar 2014 11:22:27 -0400
> From: abalas...@evaristesys.com
> To: sr-users@lists.sip-router.org
> Subject: Re: [SR-
I configured apache web server and also installed siremis 4.1.1
successfully but i couldn't login and my problem is after " welcome to
Siremis installation wizard" I went through steps 1to 4 without issue. On
the last step I even receive installation complete congratulation also
display user
IS goes to check for
> those installed components, they're not there, so it deems itself to not
> actually be installed and goes back to the installation wizard.
>
> I would venture to guess this is a web server or database permissions
> issue on your end.
>
> -- Alex
&g
Any recommendation web interface for gathering statistics other than SIREMIS? I
installed SIREMIS but unable to resolve login problem and my request for help
on ASIPTO didn't get any response also unable to find siremis mailing list.
Your help is greatly appreciated.
Thanks
Abdulmailk
I send couple of email to Asipto.com about Siremis login issue but I didn't get
any response. I configured apache web server and also installed siremis 4.1.1
successfully but i couldn't login and my problem is after " welcome to Siremis
installation wizard" I went through step 1to 4 without i
)
Hi Malik.
Kamailio is a SIP proxy it does not cater media, you can look for
Asterisk, FreeSWITCH along with Kamailio to do the needful.
On Fri, Mar 7, 2014 at 9:03 PM, malik sherif wrote:
Hello,
I am using kamailio 4.1.1, I am wondering if kamailio supports call forwarding
, three
Hello,
I am using kamailio 4.1.1, I am wondering if kamailio supports call forwarding
, three-way call, and call transfer natively? Does it expect the endpoints to
do the RTP mixing?
Thanks
Absul
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