Hi
this is Dharambir here
i created a sip trunk on CUCM 7.0 with other cluster..
i am tryinng to make call over sip trunk..
how can i trace that can show where the call is hitting..
i already used dail number analyser but not satisfying..
please
--
Regards,
Dharambir Kumar
RTMT after setting your trace level in serviceability
Sent from my iPad
On Jun 15, 2013, at 8:30 AM, Dharambir kumar varma dharambi...@gmail.com
wrote:
Hi
this is Dharambir here
i created a sip trunk on CUCM 7.0 with other cluster..
i am tryinng to make call over sip trunk..
how
Hi
please tell what parametres are required to integtate CUCM and siemens
sip trunk..
we have already configured sip trunk on both sides but one way calling
is there
from CUCM we are not able to call seimens extension
how can we trace trunk logs on rtmt please help if u can
--
Regards,
I have a question understanding the destination addresses under the SIP
Information section of the SIP Trunk Configuration. I've create 3
destination addresses going to redundant CUBE's. I understand that in
order to receive an Invite from one of the CUBE's, the address has to be
there, but
Dears,
Its worked fine with the following configuration by using the magic command
incoming uri so the dial peer will be match based on the source ip
but the problem its work for sip trunk but in case if I have h323 trunk how I
can map the customer to incoming dial-peer based on ip as the
, September 15, 2012 5:50 AM
Subject: Re: [OSL | CCIE_Voice] sip trunk
Did you configure the cucm server up
Address on the cme?
Randall
On Sep 14, 2012, at 7:32 PM, Bill Lake whl...@gmail.com wrote:
Because they are coming in on a seperate sip dial peer, but maybe that won't
work. I would
@onlinestudylist.com
Sent: Friday, September 14, 2012 6:43 AM
Subject: Re: [OSL | CCIE_Voice] sip trunk
I think cor can offer a good solution for this. Assign a separate cor group to
each dialpeer say cust1 and cust2 in both inbound and outbound directions
-Pavan
On Sep 13, 2012, at 17:45, John
:* Re: [OSL | CCIE_Voice] sip trunk
I think cor can offer a good solution for this. Assign a separate cor
group to each dialpeer say cust1 and cust2 in both inbound and outbound
directions
-Pavan
On Sep 13, 2012, at 17:45, John John john_ccie2...@yahoo.com wrote:
Dear All,
We have two sipt
ccie_voice@onlinestudylist.com
Sent: Saturday, September 15, 2012 1:11 AM
Subject: Re: [OSL | CCIE_Voice] sip trunk
Sure you can, you know that any call incoming from Site A must have ANI of
332211 and any call incoming from Site B must have 33 so you just need to
enforce that and you could
:* Saturday, September 15, 2012 1:11 AM
*Subject:* Re: [OSL | CCIE_Voice] sip trunk
Sure you can, you know that any call incoming from Site A must have ANI of
332211 and any call incoming from Site B must have 33 so you just need
to enforce that and you could do that with a incoming voice
: Bill Lake whl...@gmail.com
To: John John john_ccie2...@yahoo.com
Cc: Pavan pav.c...@gmail.com; ccie_voice@onlinestudylist.com
ccie_voice@onlinestudylist.com
Sent: Saturday, September 15, 2012 1:11 AM
Subject: Re: [OSL | CCIE_Voice] sip trunk
Sure you can, you know that any call incoming
Dear All,
We have two sipt trunk for 2 comapany:-
company A - DID range 332211XX
Company B - DID range 33XX
and each company has own PBX,Company A has AVAYA and company B has cisco Call
manager.
and they have sip trunk to my gateway where is the E1 is connected.
in my gateway there is
I think cor can offer a good solution for this. Assign a separate cor group to
each dialpeer say cust1 and cust2 in both inbound and outbound directions
-Pavan
On Sep 13, 2012, at 17:45, John John john_ccie2...@yahoo.com wrote:
Dear All,
We have two sipt trunk for 2 comapany:-
Guys,
any one done unity connection integration with SIP Trunk , i am able to
route the call to IVR but it not accepting DTMF , any clues..
Thanks Regards,
Mohammad Asad
___
For more information regarding industry leading CCIE Lab training, please
What does your sip configuration look like? You can try different DTMF
settings, sip is notorious for needing some help with the DTMF signaling, I
believe the one i used was something along the lines of nte-rtp, I'll double
check but trying looking at that!
Sent from my mobile device please
OUTPUT
!
!
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
ipv4 10.10.15.0
ipv4 10.10.16.0 255.255.240.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax
On your cube post the inbound and outbound dialpeers config. Also during a call
post the output of the command show call active voice | i Dtmf in order to
see the dtmf relay method used.
Regards,
Mohammed Al Baqari
Sent from my iPhone
On Dec 25, 2011, at 5:18 PM, Asad Yasin
Try putting the DTMF interworking commands in the dial-peer
Sent from my mobile device please excuse any typos.
On Dec 25, 2011, at 5:18 AM, Asad Yasin asad4nt...@gmail.com wrote:
OUTPUT
!
!
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
ipv4 10.10.15.0
ipv4
@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP Trunk with CUC and possible AAR
Message-ID:
CA+53e6vAJciztxfr=zerprzmboidgafrtaqr-dykezv6z5e...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Hello All,
I am working on a lab that requires to set up CUCM and CUC using SIP
Hello All,
I am working on a lab that requires to set up CUCM and CUC using SIP Trunk.
It then asks for calls that are sent to VM from BR1 to be redirected out the
PSTN when there is WAN congestion.
I have looked high and low but I can't find any reference where this can be
done with AAR...or am
Hey Inder ,
Nice catch on this , the AAR is between 2 endpoint register to the
same ccm so you cannot use the same concept of redundancy in this
config scenario , you can achieve this by having 2 route in the route
group , so the sip trunk to the CUC and also GW (of course manipulate
the
i guess you can take of the redirected number from the call and then
do mask on the VM pilot /profile or interduce tranlation pattern in
between to match the called and change the calling to 4 digits
Ash
On Thu, Oct 20, 2011 at 3:57 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
Hey Inder ,
You are right, I thought early offer when it is the case.
- Original Message -
From: Adam Frankel (afrankel) afran...@cisco.com
Date: Wednesday, June 22, 2011 12:22 am
Subject: Re: [OSL | CCIE_Voice] sip trunk delay offer
To: wormh...@sch.hu, Farkas Péter wormh...@sch.bme.hu
Cc: donny f
...@gmail.com,
ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip trunk delay offer
You are right, I thought early offer when it is the case.
- Original Message -
From: Adam Frankel (afrankel)afran...@cisco.com
Date: Wednesday, June 22, 2011 12:22 am
Subject: Re: [OSL
Have you introduce MTP to the call? By default CUCM only capable outgoing delay
offer based on G.711.
Peter
- Original Message -
From: donny f f.faraday...@gmail.com
Date: Tuesday, June 21, 2011 1:09 am
Subject: [OSL | CCIE_Voice] sip trunk delay offer
To: ccie_voice@onlinestudylist.com
...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] sip trunk delay offer
Have you introduce MTP to the call? By default CUCM only capable outgoing delay
offer based on G.711.
Peter
- Original Message -
From: donny ff.faraday...@gmail.com
Date: Tuesday, June
hi,
anybody know why i did not see the G729 in debug ? it only said PCMU in
codec
v=0
o=CiscoSystemsSIP-GW-UserAgent 6036 5234 IN IP4 10.20.100.2
s=SIP Call
c=IN IP4 10.20.100.2
t=0 0
m=audio 16522 RTP/AVP 0
c=IN IP4 10.20.100.2
a=rtpmap:0 PCMU/8000- codec
a=ptime:20
PCMU means G711U
Means the SIP message is not using G729
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of donny f
Sent: Monday, June 20, 2011 2:51 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] sip trunk delay offer
hi
Subject: [OSL | CCIE_Voice] sip trunk delay offer
hi,
anybody know why i did not see the G729 in debug ? it only said PCMU in codec
v=0
o=CiscoSystemsSIP-GW-UserAgent 6036 5234 IN IP4 10.20.100.2
s=SIP Call
c=IN IP4 10.20.100.2
t=0 0
m=audio 16522 RTP/AVP 0
c=IN IP4 10.20.100.2
a=rtpmap:0 PCMU
Thanks, that looks like a valid workaround. :)
best regards
kobel
On Sun, May 22, 2011 at 19:47, George Goglidze gogli...@gmail.com wrote:
Hi Miron,
From what I understand in your case you only have one trunk, with CUCM
Group that had two call managers.
In this case the second call
Hi,
I'm working on a lab, where I'm supposed to configure SIP trunk between HQ
CUCM and BR CUCME.
I created a SIP trunk on CUCM side with folowing CUCM group associated.
1) subscriber CUCM
2) publisher CUCM
On CUCME side I've configured to dial-peers pointing to sub with preference
0 and to
Hi Miron,
From what I understand in your case you only have one trunk, with CUCM Group
that had two call managers.
In this case the second call manager is only used in case if the first fails,
so it will not work as CUCME can't get to the Publisher.
If you want to cover the failover in the
Hi all.
I am setting up a sip trunk with a local ITSP. Now they havent provided me
with
any username and password. Can someone tell me what configuration do i need to
do on router ?
My router is 2821 with 12.4(22)T adventerprise
They have just provided me the sip server address and range
Got CUCM 8.0 configured with a SIP trunk for PSTN calls. I've got Media
Termination Point Required enabled on the trunk but I've confirmed that CM
is not sending any SDP information to the PSTN, which is what they are
expecting. What am I missing?
Thanks,
Mark
You will only get SDP if you enable SIP early offer.
Sent from my iPhone
11 feb 2011 kl. 18:56 skrev ccie_voice-requ...@onlinestudylist.com
ccie_voice-requ...@onlinestudylist.com:
Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestudylist.com
To subscribe or unsubscribe
)
--
Message: 1
Date: Fri, 11 Feb 2011 11:16:26 -0500
From: Mark Davis davismar...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP trunk to PSTN - CM 8.0 not sending SDP
header in SIP INVITE packet
Message-ID:
AANLkTikKe4nMaV_R2ONAEMMx=vakanq9cxzqv1bmd
...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP trunk to PSTN - CM 8.0 not sending SDP
header in SIP INVITE packet
Message-ID:
AANLkTikKe4nMaV_R2ONAEMMx=vakanq9cxzqv1bmd...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Got CUCM 8.0 configured
header in SIP
INVITE packet (Mark Davis)
--
Message: 1
Date: Fri, 11 Feb 2011 11:16:26 -0500
From: Mark Davis davismar...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP trunk to PSTN
| CCIE_Voice] SIP trunk to PSTN - CM 8.0 not sending SDP
header in SIP INVITE packet
Message-ID:
AANLkTikKe4nMaV_R2ONAEMMx=vakanq9cxzqv1bmd...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Got CUCM 8.0 configured with a SIP trunk for PSTN calls. I've got Media
Termination
Hi:
I have a sip trunk to my pstn router I'm trying to check the codec that the
call is using but I can't this info at ucm traces or pstn gw debugs.
I have try sip stack traces at ucm and also deb ccsip all at pstn, but I can't
this info
Any suggestion?
You should try debug ccsip messages on the PSTN or CUBE router. It will
show you the codec negotiation.
On May 29, 2010 1:55 PM, Angel Perez gorr...@hotmail.com wrote:
Hi:
I have a sip trunk to my pstn router I'm trying to check the codec that the
call is using but I can't this info at ucm
Yes you should pick it up in the invite and OK messages thus
m=audio 47100 RTP/AVP 8 0 18 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:98 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
Thanks, is it possible to check the call type also?
Regards
Subject: Re: [OSL | CCIE_Voice] SIP TRUNK
From: ghopk...@wolf-rock.co.uk
Date: Sat, 29 May 2010 20:04:36 +0100
CC: ccie_voice@onlinestudylist.com
To: gorr...@hotmail.com
Yes you should pick it up in the invite and OK messages
Thanks Arun that helps.
-AK
On Thu, Jan 21, 2010 at 9:00 PM, Arun Kumar arunv...@gmail.com wrote:
if you have the gateways/trunk registered to the CUCM and it is not showing
when you want to do route group
it means you have used them directly under the route pattern.
On Fri, Jan 22, 2010
Hi
I'm working on Lab5 in Vol1.
I noticed wired issue, when I tried to create a Route Group it's not showing
any Devices(GW/Trunk) so I removed them and added manually but some how it's
not showing my sip trunk and GK trunk while I'm adding route pattern, tried
add / delete couple of times but
since that device is assigned to a RP directly remove that RP and add the
device in a RG check the dependacy of the device so u will know on which RP
u assigned it
On Thu, Jan 21, 2010 at 9:59 AM, Arun Kumar arunv...@gmail.com wrote:
Hi
I'm working on Lab5 in Vol1.
I noticed wired issue,
Hi Mill,
got this working, anyway thanks very much for your help.
Thanks
Arun
On Fri, Jan 22, 2010 at 1:25 AM, kill mill jha...@gmail.com wrote:
since that device is assigned to a RP directly remove that RP and add the
device in a RG check the dependacy of the device so u will know on which
So Arun how did you make it work please, I am having same issue too. Thanks.
On Thu, Jan 21, 2010 at 11:55 AM, kill mill jha...@gmail.com wrote:
since that device is assigned to a RP directly remove that RP and add the
device in a RG check the dependacy of the device so u will know on which RP
Lab , CCIE Voice Lab and CCIE Storage
Lab Certifications.
From: Hawkins Jason L NGA-ES USA CTR jason.l.hawkins@nga.mil
Date: Mon, 9 Nov 2009 21:35:00 -0500
To: OSL Group ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP TRUNK VOL2 LAB5 q 2.10
The question asks us
The question asks us to route all international calls over a SIP trunk
to a SIP ITSP. One of the requirements is to mark both the ANI and
DNIS as type International. I didn't think SIP trunks supported number
type. Is there a reason for setting the number type if it isn't
supported on the
Check out the command sip-ua in global config. Your username and
password will go there, along with some other settings...
Sent while mobile.
On Oct 28, 2009, at 21:20, J Hogan j.jho...@gmail.com wrote:
I have a SIp trunk provider offerin me 2 SIP trunk Likes and there
is a requirment for a
I wanted to do a SIP trunk directly from CM to the SIP provider and not On
the router.
On Thu, Oct 29, 2009 at 8:21 AM, Jason Granat j...@slash128.com wrote:
Check out the command sip-ua in global config. Your username and
password will go there, along with some other settings...
Sent while
, 2009 6:44 AM
To: Jason Granat
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SIP trunk
I wanted to do a SIP trunk directly from CM to the SIP provider and not On the
router.
On Thu, Oct 29, 2009 at 8:21 AM, Jason Granat
j...@slash128.commailto:j...@slash128.com wrote:
Check out
for on CUCM:
https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/7_0_1/secugd/secrealm.html
*From:* J Hogan [mailto:j.jho...@gmail.com]
*Sent:* Thursday, October 29, 2009 6:44 AM
*To:* Jason Granat
*Cc:* ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] SIP trunk
)
--
Message: 1
Date: Thu, 29 Oct 2009 10:51:12 -0500
From: J Hogan j.jho...@gmail.com
Subject: Re: [OSL | CCIE_Voice] SIP trunk
To: Jason Granat j...@slash128.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Message-ID
I have a SIp trunk provider offerin me 2 SIP trunk Likes and there is a
requirment for a USER ID and password But I do not see where to inter
that does anyone have some documentation somewere on how to et this
type of SIP trunk to work?
--
J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI
behavior.
From: mciarfe...@iplogic.com
To: spoli...@hotmail.com; ccie_voice@onlinestudylist.com
Date: Mon, 28 Sep 2009 23:00:31 -0400
Subject: RE: [OSL | CCIE_Voice] SIP Trunk Blocking Calling Name
Try configuring it on the dial-peer.
From: ccie_voice-boun...@onlinestudylist.com
| CCIE_Voice] SIP Trunk Blocking Calling Name
Hi,
I'm trying to block just the calling name via SIP trunk but when I set Calling
Name Presentation to restricted it block the number also. For MGCP and H323
Trunks, no problems.
Something that is missing?
Thanks in advance, Sergio
Hi,
I'm trying to block just the calling name via SIP trunk but when I set Calling
Name Presentation to restricted it block the number also. For MGCP and H323
Trunks, no problems.
Something that is missing?
Thanks in advance, Sergio.
I think I understand this but just for clarification I have a few questions
If I configured a SIP trunk between CM (version 4) and the HQ-RTR:
1) I should force G711 only in both directions across the trunk using
inbound and outbound dial-peers (codec g711ulaw) on the HQ-RTR
2) I
. Faster and quicker if you preplan.
- Original Message -
*From:* Mike Brooks 2xcci...@gmail.com
*To:* ccie_voice@onlinestudylist.com
*Cc:* bryan.d.bro...@gmail.com
*Sent:* Sunday, May 03, 2009 3:04 PM
*Subject:* [OSL | CCIE_Voice] SIP Trunk G711
I think I understand this but just
-
From: Mike Brooks
To: Afatsum ; OSL Group
Cc: bryan.d.bro...@gmail.com
Sent: Sunday, May 03, 2009 3:40 PM
Subject: Re: [OSL | CCIE_Voice] SIP Trunk G711
This is how I have it set up. This allows the codec to remain G729 across
the WAN. Also the HQ-MRGL in the HQ-DP contains
a separate region, DP,
MRG, MRGL etc for SIP trunk. Faster and quicker if you preplan.
- Original Message -
*From:* Mike Brooks 2xcci...@gmail.com
*To:* ccie_voice@onlinestudylist.com
*Cc:* bryan.d.bro...@gmail.com
*Sent:* Sunday, May 03, 2009 3:04 PM
*Subject:* [OSL | CCIE_Voice] SIP Trunk
Subject: Re: [OSL | CCIE_Voice] SIP Trunk G711
Oh, I see your point.
So the SIPTRUNK-MRGL shoud have ONLY the unicast-moh server, software MTP,
and 6608 transcoder ?
What about the other resources such as 6608 conference or software
conferenceor annunciator ? Shouldn't these be included
And I think SW MTP shd be in G711 only DP too.
--- On Mon, 5/4/09, Afatsum afat...@verizon.net wrote:
From: Afatsum afat...@verizon.net
Subject: Re: [OSL | CCIE_Voice] SIP Trunk G711
To: Mike Brooks 2xcci...@gmail.com
Cc: OSL Group ccie_voice@onlinestudylist.com, bryan.d.bro...@gmail.com
Date
Here's a good one! When I make a call over a sip trunk and a sccp phone
picks up and performs a blind transfer to another sccp phone, there is no
ringback heard on the calling phone. no MTP or annunciator is configured
on the UCM cluster. Any idea where to start? I suspect UCM is not
providing
| CCIE_Voice] SIP Trunk Issue
Date: Wed, 1 Apr 2009 23:16:14 -0500
Here's a good one! When I make a call over a sip trunk and a sccp
phone picks up and performs a blind transfer to another sccp phone,
there is no ringback heard on the calling phone. no MTP or
annunciator is configured
Hi List,
last night i faced a problem with a SIP-Trunk from CCM to CCME. On the
Call-Flow PHONE-1-CCM-SIP-Trunk-CCME-PHONE-2 i could only get the first
DTMF-Tone, from Phone-1, through any subsequent DTMFs were not received by
Phone-2. DTMF Tones from PHONE-2 to PHONE-1 were working just fine.
Lab Certifications.
From: Robert Schuknecht rschukne...@gmx.de
Date: Fri, 20 Feb 2009 12:06:46 +0100
To: OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP-Trunk: No DTMF
Hi List,
last night i faced a problem with a SIP-Trunk from CCM to CCME
Call drom CCM to CME phone via SIP trunk works great, but when the CME
phone Forwards to CUE, the CME phone stops ringing, the CCM phone
continues to ring forever (No transfer to Users VM)...Any ideas?
...@netelligent.com
Date: Mon, 12 Jan 2009 11:41:43 -0600
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP Trunk transfer to CUE
Call drom CCM to CME phone via SIP trunk works great, but when the CME
phone Forwards to CUE, the CME phone stops ringing, the CCM phone
continues to ring
.
From: Kevin Porter kpor...@netelligent.com
Date: Mon, 12 Jan 2009 11:41:43 -0600
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP Trunk transfer to CUE
Call drom CCM to CME phone via SIP trunk works great, but when the CME
phone Forwards to CUE, the CME phone stops
1. Can Gatekeeper, H323 vgw and SIP vgw co-exist?
2. From CCM site route PSTN outgoing call via SIP trunk, in SIP VGW, are the
below configs enough?
voice service voip
sip
bind all source fa0/0.100
sip-ua
retry timer 3600
dial-peer voice 9 pots-- For all outbound PSTN call via E1
Hi there,
ASCII diagram:
CCM SIP trunkBR2CUE
Has anyone seen this before?
Incoming dial-peer on BR2:
dial-peer voice 3200 voip
destination-pattern [12]...$
session protocol sipv2
session target ipv4:172.1.200.1
incoming called-number 3[12]00$
dtmf-relay sip-notify rtp-nte
Allright, found the answer myself after ~1.5 hours of trying:
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html#wp1342177
rtp-nte to sip-notify seems to be not supported directly.
However, if recirculating the call via H.323 on CME itself the
Are these accurate statements?
1.A SIP trunk will always require unicast MOH,
regardless of whether it's G.711 (to CUE site) or G.729.
2.A SIP trunk's MTP, regardless off software or hardware
(transcoder), will always require unicast MOH.
What could
| CCIE_Voice] SIP trunk between CME and CCM 4.1
Hi,
I am trying a topology sip trunk, with ccm 4.1 and CME, CME source ip
address - 172.16.2.100, from CME IPPhone I am able to reach CCM IPPhone, but
from CCM IPPhone to CME I am not able to reach CME IPPhone.
I am having route pattern in CCM
Hi,
I am trying a topology sip trunk, with ccm 4.1 and CME, CME source ip address
- 172.16.2.100, from CME IPPhone I am able to reach CCM IPPhone, but from CCM
IPPhone to CME I am not able to reach CME IPPhone.
I am having route pattern in CCM to CME as siptrunk as gateway,
In
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