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<P><FONT SIZE=2>-----Original Message-----<BR>
From: cisco-voip-bounces@puck.nether.net <cisco-voip-bounces@puck.nether.net><BR>
To: cisco-voip@puck.nether.net <cisco-voip@puck.nether.net><BR>
Sent: Sat Mar 26 05:00:01 2011<BR>
Subject: cisco-voip Digest, Vol 89, Issue 25<BR>
<BR>
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<BR>
Today's Topics:<BR>
<BR>
1. Easy On Hold Music question (Mike King)<BR>
2. Re: Easy On Hold Music question (Jobe Gates)<BR>
3. Re: Easy On Hold Music question (Nate VanMaren)<BR>
4. Re: alarm panel issues and VG224 w/ SCCP (Fuermann, Jason)<BR>
5. Re: Possible to send Pre-recorded Message to Distribution<BR>
list-Unity 7? (Mark Pratt)<BR>
6. Re: Easy On Hold Music question (Dennis Heim)<BR>
7. Messages Softkey removal on 7911G (Dan Greenway)<BR>
8. Re: alarm panel issues and VG224 w/ SCCP (Lelio Fulgenzi)<BR>
9. Re: alarm panel issues and VG224 w/ SCCP (Nick Matthews)<BR>
10. Re: alarm panel issues and VG224 w/ SCCP (Lelio Fulgenzi)<BR>
11. SIP trunk configuration ! (Cisco Voip)<BR>
12. Re: SIP trunk configuration ! (Paul)<BR>
13. Re: SIP trunk configuration ! (Cisco Voip)<BR>
14. Re: SIP trunk configuration ! (Michel L. M. B. Perez)<BR>
15. Re: SIP trunk configuration ! (Mark Holloway)<BR>
<BR>
<BR>
----------------------------------------------------------------------<BR>
<BR>
Message: 1<BR>
Date: Thu, 24 Mar 2011 12:15:26 -0400<BR>
From: Mike King <me@mpking.com><BR>
To: Cisco VoIPoE List <cisco-voip@puck.nether.net><BR>
Subject: [cisco-voip] Easy On Hold Music question<BR>
Message-ID:<BR>
<AANLkTik527spJ05tOgEJwK7x4g=pyd97VqVb_3Q1UV_A@mail.gmail.com><BR>
Content-Type: text/plain; charset="iso-8859-1"<BR>
<BR>
We so have a new marketing person, and they want to replace the On hold<BR>
music with advertising. Currently the Same On hold music is applied to<BR>
everyone.<BR>
<BR>
We have around 20 different departments, and each one wants to have<BR>
different Advertising (We'll have an outside company prepare the Hold music<BR>
with the wording we want, They can output any codec we want)<BR>
<BR>
We currently have 1 Pub, and 2 Subs (and a bunch of remote sites that have<BR>
SRST routers that are 2921's) no Multicast between sites.<BR>
<BR>
So is this possible (I'm 99% positive it is, but I don't want to go into the<BR>
meeting without being 100%)<BR>
<BR>
Is this a good idea? (Technical standpoint)<BR>
<BR>
I'm trying to make sure they're isn't a limit like you can only have 5 On<BR>
hold music sources or only 10 Device Pools something.<BR>
<BR>
If i remember correctly, MOH can be set Per Device Pool (although it didn't<BR>
seem readily apparent there) Per device, or per Line.<BR>
<BR>
Any opinions are being sought as well.<BR>
<BR>
Mike<BR>
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Message: 2<BR>
Date: Thu, 24 Mar 2011 12:22:46 -0400<BR>
From: Jobe Gates <jobe@gates-tribe.com><BR>
To: Mike King <me@mpking.com><BR>
Cc: Cisco VoIPoE List <cisco-voip@puck.nether.net><BR>
Subject: Re: [cisco-voip] Easy On Hold Music question<BR>
Message-ID: <0284839B-4225-43D7-B023-2F4B62D1C7BF@gates-tribe.com><BR>
Content-Type: text/plain; charset=us-ascii<BR>
<BR>
I think the limit is 50 MoH sources. I had 30 locations and each one wanted different MoH. Worked fine. <BR>
<BR>
Thanks,<BR>
Jobe<BR>
<BR>
On Mar 24, 2011, at 12:15 PM, Mike King <me@mpking.com> wrote:<BR>
<BR>
> We so have a new marketing person, and they want to replace the On hold music with advertising. Currently the Same On hold music is applied to everyone.<BR>
><BR>
> We have around 20 different departments, and each one wants to have different Advertising (We'll have an outside company prepare the Hold music with the wording we want, They can output any codec we want)<BR>
><BR>
> We currently have 1 Pub, and 2 Subs (and a bunch of remote sites that have SRST routers that are 2921's) no Multicast between sites.<BR>
><BR>
> So is this possible (I'm 99% positive it is, but I don't want to go into the meeting without being 100%)<BR>
><BR>
> Is this a good idea? (Technical standpoint)<BR>
><BR>
> I'm trying to make sure they're isn't a limit like you can only have 5 On hold music sources or only 10 Device Pools something.<BR>
><BR>
> If i remember correctly, MOH can be set Per Device Pool (although it didn't seem readily apparent there) Per device, or per Line.<BR>
><BR>
> Any opinions are being sought as well.<BR>
><BR>
> Mike<BR>
><BR>
><BR>
><BR>
> _______________________________________________<BR>
> cisco-voip mailing list<BR>
> cisco-voip@puck.nether.net<BR>
> <A HREF="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</A><BR>
<BR>
<BR>
<BR>
------------------------------<BR>
<BR>
Message: 3<BR>
Date: Thu, 24 Mar 2011 16:24:43 +0000<BR>
From: Nate VanMaren <VanMarenNP@ldschurch.org><BR>
To: Mike King <me@mpking.com>, Cisco VoIPoE List<BR>
<cisco-voip@puck.nether.net><BR>
Subject: Re: [cisco-voip] Easy On Hold Music question<BR>
Message-ID:<BR>
<2F143E71016CA34C924BF4C33AEF21100D74F1@W1185.ldschurch.org><BR>
Content-Type: text/plain; charset="iso-8859-1"<BR>
<BR>
SRST used to be only able to send one file out. Now it can do 5.<BR>
<A HREF="http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/feature/guide/MOH_srst.html">http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/feature/guide/MOH_srst.html</A><BR>
<BR>
Are you streaming unicast MOH across the WAN?<BR>
<BR>
<BR>
From: cisco-voip-bounces@puck.nether.net [<A HREF="mailto:cisco-voip-bounces@puck.nether.net">mailto:cisco-voip-bounces@puck.nether.net</A>] On Behalf Of Mike King<BR>
Sent: Thursday, March 24, 2011 10:15 AM<BR>
To: Cisco VoIPoE List<BR>
Subject: [cisco-voip] Easy On Hold Music question<BR>
<BR>
We so have a new marketing person, and they want to replace the On hold music with advertising. Currently the Same On hold music is applied to everyone.<BR>
<BR>
We have around 20 different departments, and each one wants to have different Advertising (We'll have an outside company prepare the Hold music with the wording we want, They can output any codec we want)<BR>
<BR>
We currently have 1 Pub, and 2 Subs (and a bunch of remote sites that have SRST routers that are 2921's) no Multicast between sites.<BR>
<BR>
So is this possible (I'm 99% positive it is, but I don't want to go into the meeting without being 100%)<BR>
<BR>
Is this a good idea? (Technical standpoint)<BR>
<BR>
I'm trying to make sure they're isn't a limit like you can only have 5 On hold music sources or only 10 Device Pools something.<BR>
<BR>
If i remember correctly, MOH can be set Per Device Pool (although it didn't seem readily apparent there) Per device, or per Line.<BR>
<BR>
Any opinions are being sought as well.<BR>
<BR>
Mike<BR>
<BR>
<BR>
<BR>
<BR>
NOTICE: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message.<BR>
<BR>
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<BR>
Message: 4<BR>
Date: Thu, 24 Mar 2011 11:46:34 -0500<BR>
From: "Fuermann, Jason" <jason.f@shsu.edu><BR>
To: "'Lelio Fulgenzi'" <lelio@uoguelph.ca>,<BR>
"cisco-voip@puck.nether.net" <cisco-voip@puck.nether.net><BR>
Subject: Re: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
Message-ID:<BR>
<8FAC1E47484E43469AA28DBF35C955E4D8477C8806@EXMBX.SHSU.EDU><BR>
Content-Type: text/plain; charset="utf-8"<BR>
<BR>
The head end is probably hanging up while the dialer is still spitting out codes, and the gw thinks it?s trying to dial an extension. Just a guess.<BR>
<A HREF="http://markmail.org/message/kwzlalbdo2asbxqo">http://markmail.org/message/kwzlalbdo2asbxqo</A><BR>
<BR>
<BR>
From: cisco-voip-bounces@puck.nether.net [<A HREF="mailto:cisco-voip-bounces@puck.nether.net">mailto:cisco-voip-bounces@puck.nether.net</A>] On Behalf Of Lelio Fulgenzi<BR>
Sent: Thursday, March 24, 2011 10:55 AM<BR>
To: cisco-voip@puck.nether.net<BR>
Subject: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
<BR>
We have a remote site which uses alarms on their VG224. It seems they're not working properly. I've got a call back into the client to arrange a conference call with their alarm provider so I have access to technical details I can share.<BR>
<BR>
When I look at the CDRs, I see some strange entries. Basically a call to a toll free number, followed by short calls of 1 second with invalid DNs. (see below)<BR>
<BR>
I recall some talk about alarms and SCCP and it's better with MGCP. Can anyone who had this problem confirm if this was the same issue?<BR>
<BR>
I have no problem setting up a few ports for MGCP but I have some concerns:<BR>
<BR>
* can I have a VG224 with some ports as SCCP and some as MGCP?<BR>
* how can I ensure that my MGCP ports work during failover, i.e. SRST?<BR>
<BR>
* I don't want to have to redo all my dial-peers. I don't mind putting in one dial that sends all calls from the VG224 to the SRST router, but I'd like an inbound dialpeer from the VG224 that assigns the same class of service for all inbound calls from that vg224.<BR>
* Is this doable?<BR>
<BR>
________________________________<BR>
call 1<BR>
<BR>
call 2<BR>
<BR>
DateTimeOrigination<BR>
<BR>
DateTimeOrigination<BR>
<BR>
Mar 8, 2011 10:02:10 AM<BR>
<BR>
Mar 8, 2011 10:02:34 AM<BR>
<BR>
OriginalCalledPartyNumber<BR>
<BR>
OriginalCalledPartyNumber<BR>
<BR>
1866------- <snip><BR>
<BR>
7000C<BR>
<BR>
FinalCalledPartyNumber<BR>
<BR>
FinalCalledPartyNumber<BR>
<BR>
1866------- <snip><BR>
<BR>
7000C<BR>
<BR>
DateTimeConnect<BR>
<BR>
DateTimeConnect<BR>
<BR>
Mar 8, 2011 10:02:22 AM<BR>
<BR>
N/A<BR>
<BR>
DateTimeDisconnect<BR>
<BR>
DateTimeDisconnect<BR>
<BR>
Mar 8, 2011 10:02:34 AM<BR>
<BR>
Mar 8, 2011 10:02:36 AM<BR>
<BR>
Duration<BR>
<BR>
Duration<BR>
<BR>
12<BR>
<BR>
0<BR>
<BR>
________________________________<BR>
<BR>
---<BR>
Lelio Fulgenzi, B.A.<BR>
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<BR>
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<BR>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>
Cooking with unix is easy. You just sed it and forget it.<BR>
- LFJ (with apologies to Mr. Popeil)<BR>
<BR>
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<BR>
Message: 5<BR>
Date: Thu, 24 Mar 2011 11:01:11 -0700<BR>
From: Mark Pratt <Mark.Pratt@wageworks.com><BR>
To: Cisco VoIPoE List <cisco-voip@puck.nether.net><BR>
Subject: Re: [cisco-voip] Possible to send Pre-recorded Message to<BR>
Distribution list-Unity 7?<BR>
Message-ID:<BR>
<8E7333F435A1BD48917AD112B1EDB0D50BEC3EFD5B@wwexmbxp01.wageworks.local><BR>
<BR>
Content-Type: text/plain; charset="us-ascii"<BR>
<BR>
Thanks for all the suggestions. I think what Erik suggested is what we will be going with, but I do plan on exploring the other options suggested for future use.<BR>
<BR>
Thanks,<BR>
Mark<BR>
(480) 291-0431 or x40431<BR>
<BR>
From: Erick B. [<A HREF="mailto:erickbee@gmail.com">mailto:erickbee@gmail.com</A>]<BR>
Sent: Tuesday, March 22, 2011 10:50 PM<BR>
To: Mark Pratt<BR>
Cc: Cisco VoIPoE List<BR>
Subject: Re: [cisco-voip] Possible to send Pre-recorded Message to Distribution list-Unity 7?<BR>
<BR>
Not sure if it is possible with a distribution list in Unity, but there is Broadcast Messages, and there is a tool on unitytools.com<<A HREF="http://unitytools.com">http://unitytools.com</A>> to manage broadcast messages already on the server to send them out again, etc.<BR>
<BR>
Maybe you can leverage this... maybe send it to yourself then forward it to a distribution list if broadcast messages isn't what you need. Of all the folks I've worked on, only one has used the broadcast message feature a few times.<BR>
<BR>
<A HREF="http://www.ciscounitytools.com/Applications/Unity/BroadcastMessageManager/BroadcastMessageManager.html">http://www.ciscounitytools.com/Applications/Unity/BroadcastMessageManager/BroadcastMessageManager.html</A><BR>
<BR>
<BR>
On Tue, Mar 22, 2011 at 11:20 AM, Mark Pratt <Mark.Pratt@wageworks.com<<A HREF="mailto:Mark.Pratt@wageworks.com">mailto:Mark.Pratt@wageworks.com</A>>> wrote:<BR>
Does anyone know if it is possible to send a pre-recorded message to a Distribution list in Unity 7, and if so how to do it? My company is looking into sending out periodic announcements via voicemail companywide and some of the messages would be pre-recorded.<BR>
<BR>
Thanks in advance.<BR>
Mark<BR>
<BR>
Mark Pratt<BR>
Telecom Engineer, IT Operations<BR>
Direct: 480.291.0431<tel:480.291.0431> | Mobile: 602.284.3568<tel:602.284.3568> |<BR>
Email: Mark.Pratt@wageworks.com<<A HREF="mailto:Mark.pratt@wageworks.com">mailto:Mark.pratt@wageworks.com</A>><BR>
<BR>
[cid:image001.jpg@01CBEA12.BD62F0C0]<<A HREF="http://www.wageworks.com/">http://www.wageworks.com/</A>><BR>
<BR>
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<BR>
Message: 6<BR>
Date: Thu, 24 Mar 2011 18:21:32 +0000<BR>
From: Dennis Heim <Dennis.Heim@cdw.com><BR>
To: Nate VanMaren <VanMarenNP@ldschurch.org>, Mike King<BR>
<me@mpking.com>, Cisco VoIPoE List <cisco-voip@puck.nether.net><BR>
Subject: Re: [cisco-voip] Easy On Hold Music question<BR>
Message-ID:<BR>
<7BA674F8A3CAAF4FADB174DB8C255E8A023072@EXMBSW2VH.corp.cdw.com><BR>
Content-Type: text/plain; charset="us-ascii"<BR>
<BR>
You probably need to stream multicast moh from each sites router. It will work based on the info you provided. However, from a management perspective it can be a real pain to have to update all those router's flash with the new file.<BR>
<BR>
Dennis Heim<BR>
Network Voice Engineer<BR>
CDW Advanced Technology Services<BR>
11711 N. Meridian Street, Suite 225<BR>
Carmel, IN 46032<BR>
<BR>
317.569.4255 Single Number Reach<BR>
317.569.4201 Fax<BR>
dennis.heim@cdw.com<<A HREF="mailto:dennis.heim@cdw.com">mailto:dennis.heim@cdw.com</A>><BR>
cdw.com/content/solutions/unified-communications/<<A HREF="http://www.cdw.com/content/solutions/unified-communications/">http://www.cdw.com/content/solutions/unified-communications/</A>><BR>
<BR>
From: cisco-voip-bounces@puck.nether.net [<A HREF="mailto:cisco-voip-bounces@puck.nether.net">mailto:cisco-voip-bounces@puck.nether.net</A>] On Behalf Of Nate VanMaren<BR>
Sent: Thursday, March 24, 2011 12:25 PM<BR>
To: Mike King; Cisco VoIPoE List<BR>
Subject: Re: [cisco-voip] Easy On Hold Music question<BR>
<BR>
SRST used to be only able to send one file out. Now it can do 5.<BR>
<A HREF="http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/feature/guide/MOH_srst.html">http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/feature/guide/MOH_srst.html</A><BR>
<BR>
Are you streaming unicast MOH across the WAN?<BR>
<BR>
<BR>
From: cisco-voip-bounces@puck.nether.net [<A HREF="mailto:cisco-voip-bounces@puck.nether.net">mailto:cisco-voip-bounces@puck.nether.net</A>] On Behalf Of Mike King<BR>
Sent: Thursday, March 24, 2011 10:15 AM<BR>
To: Cisco VoIPoE List<BR>
Subject: [cisco-voip] Easy On Hold Music question<BR>
<BR>
We so have a new marketing person, and they want to replace the On hold music with advertising. Currently the Same On hold music is applied to everyone.<BR>
<BR>
We have around 20 different departments, and each one wants to have different Advertising (We'll have an outside company prepare the Hold music with the wording we want, They can output any codec we want)<BR>
<BR>
We currently have 1 Pub, and 2 Subs (and a bunch of remote sites that have SRST routers that are 2921's) no Multicast between sites.<BR>
<BR>
So is this possible (I'm 99% positive it is, but I don't want to go into the meeting without being 100%)<BR>
<BR>
Is this a good idea? (Technical standpoint)<BR>
<BR>
I'm trying to make sure they're isn't a limit like you can only have 5 On hold music sources or only 10 Device Pools something.<BR>
<BR>
If i remember correctly, MOH can be set Per Device Pool (although it didn't seem readily apparent there) Per device, or per Line.<BR>
<BR>
Any opinions are being sought as well.<BR>
<BR>
Mike<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
NOTICE: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message.<BR>
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<BR>
Message: 7<BR>
Date: Thu, 24 Mar 2011 16:53:37 +0000<BR>
From: Dan Greenway <Dan.Greenway@2e2.com><BR>
To: "cisco-voip@puck.nether.net" <cisco-voip@puck.nether.net><BR>
Subject: [cisco-voip] Messages Softkey removal on 7911G<BR>
Message-ID:<BR>
<3D08D6418EAD5E4C99803C382EA40B3A2368851B79@JESS.prime-uk.local><BR>
Content-Type: text/plain; charset="us-ascii"<BR>
<BR>
Does anyone know of a way to remove the messages softkey on a 7911G? We are running CUCM8.5 with no VM so would like it removed.<BR>
<BR>
Thanks<BR>
Dan<BR>
<BR>
<BR>
<BR>
<BR>
Daniel Greenway<BR>
<BR>
UC Engineer<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
E<BR>
<BR>
dan.greenway@2e2.com<BR>
<BR>
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2e2 is an ICT Lifecycle Services Provider. To find out more visit www.2e2.com<<A HREF="http://www.2e2.com/">http://www.2e2.com/</A>><BR>
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<BR>
Message: 8<BR>
Date: Thu, 24 Mar 2011 14:40:28 -0400 (EDT)<BR>
From: Lelio Fulgenzi <lelio@uoguelph.ca><BR>
To: Jason Fuermann <jason.f@shsu.edu><BR>
Cc: cisco-voip@puck.nether.net<BR>
Subject: Re: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
Message-ID:<BR>
<325586484.2842634.1300992028448.JavaMail.root@simcoe.cs.uoguelph.ca><BR>
Content-Type: text/plain; charset="utf-8"<BR>
<BR>
That was my suspicion. Just wondering if those who went from SCCP to MGCP/H323 were encountering the same thing.<BR>
<BR>
We'll have to see what the alarm vendor says.<BR>
<BR>
I'm hoping someone on the list who has sccp/h323 configured ports on a vg224 can help with my other questions.<BR>
<BR>
---<BR>
Lelio Fulgenzi, B.A.<BR>
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<BR>
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<BR>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>
Cooking with unix is easy. You just sed it and forget it.<BR>
- LFJ (with apologies to Mr. Popeil)<BR>
<BR>
<BR>
----- Original Message -----<BR>
From: "Jason Fuermann" <jason.f@shsu.edu><BR>
To: "Lelio Fulgenzi" <lelio@uoguelph.ca>, cisco-voip@puck.nether.net<BR>
Sent: Thursday, March 24, 2011 12:46:34 PM<BR>
Subject: RE: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
<BR>
<BR>
<BR>
<BR>
The head end is probably hanging up while the dialer is still spitting out codes, and the gw thinks it?s trying to dial an extension. Just a guess.<BR>
<BR>
<A HREF="http://markmail.org/message/kwzlalbdo2asbxqo">http://markmail.org/message/kwzlalbdo2asbxqo</A><BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
From: cisco-voip-bounces@puck.nether.net [<A HREF="mailto:cisco-voip-bounces@puck.nether.net">mailto:cisco-voip-bounces@puck.nether.net</A>] On Behalf Of Lelio Fulgenzi<BR>
Sent: Thursday, March 24, 2011 10:55 AM<BR>
To: cisco-voip@puck.nether.net<BR>
Subject: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
<BR>
<BR>
<BR>
<BR>
We have a remote site which uses alarms on their VG224. It seems they're not working properly. I've got a call back into the client to arrange a conference call with their alarm provider so I have access to technical details I can share.<BR>
<BR>
When I look at the CDRs, I see some strange entries. Basically a call to a toll free number, followed by short calls of 1 second with invalid DNs. (see below)<BR>
<BR>
I recall some talk about alarms and SCCP and it's better with MGCP. Can anyone who had this problem confirm if this was the same issue?<BR>
<BR>
I have no problem setting up a few ports for MGCP but I have some concerns:<BR>
<BR>
? can I have a VG224 with some ports as SCCP and some as MGCP?<BR>
? how can I ensure that my MGCP ports work during failover, i.e. SRST?<BR>
<BR>
<BR>
<BR>
<BR>
?<BR>
I don't want to have to redo all my dial-peers. I don't mind putting in one dial that sends all calls from the VG224 to the SRST router, but I'd like an inbound dialpeer from the VG224 that assigns the same class of service for all inbound calls from that vg224.<BR>
? Is this doable?<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
call 1 <BR>
<BR>
call 2<BR>
<BR>
<BR>
DateTimeOrigination <BR>
<BR>
DateTimeOrigination<BR>
<BR>
<BR>
Mar 8, 2011 10:02:10 AM <BR>
<BR>
Mar 8, 2011 10:02:34 AM<BR>
<BR>
<BR>
OriginalCalledPartyNumber <BR>
<BR>
OriginalCalledPartyNumber<BR>
<BR>
<BR>
1866------- <snip> <BR>
<BR>
7000C<BR>
<BR>
<BR>
FinalCalledPartyNumber <BR>
<BR>
FinalCalledPartyNumber<BR>
<BR>
<BR>
1866------- <snip> <BR>
<BR>
7000C<BR>
<BR>
<BR>
DateTimeConnect <BR>
<BR>
DateTimeConnect<BR>
<BR>
<BR>
Mar 8, 2011 10:02:22 AM <BR>
<BR>
N/A<BR>
<BR>
<BR>
DateTimeDisconnect <BR>
<BR>
DateTimeDisconnect<BR>
<BR>
<BR>
Mar 8, 2011 10:02:34 AM <BR>
<BR>
Mar 8, 2011 10:02:36 AM<BR>
<BR>
<BR>
Duration <BR>
<BR>
Duration<BR>
<BR>
<BR>
12 <BR>
<BR>
0<BR>
<BR>
<BR>
<BR>
<BR>
---<BR>
Lelio Fulgenzi, B.A.<BR>
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<BR>
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<BR>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>
Cooking with unix is easy. You just sed it and forget it.<BR>
- LFJ (with apologies to Mr. Popeil)<BR>
<BR>
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<BR>
------------------------------<BR>
<BR>
Message: 9<BR>
Date: Thu, 24 Mar 2011 18:08:41 -0400<BR>
From: Nick Matthews <matthnick@gmail.com><BR>
To: Lelio Fulgenzi <lelio@uoguelph.ca><BR>
Cc: cisco-voip@puck.nether.net<BR>
Subject: Re: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
Message-ID:<BR>
<AANLkTinmyN8EUS7Ztp92=MWDiLnCMXN8cret6Z6w7cFi@mail.gmail.com><BR>
Content-Type: text/plain; charset="windows-1252"<BR>
<BR>
Can almost tell you the problem from the title of the email:<BR>
<BR>
The problem with using MGCP/SCCP on these ports is that they are going to do<BR>
DTMF relay. They will take the TDM tone, convert it to sccp, and replay it<BR>
on the other side. This doesn't jive with alarm panels because their timing<BR>
is very specific. They basically use DTMF tones like modem tones and send<BR>
information back and forth via DTMF. Because SCCP DTMF messages have a fixed<BR>
duration when they are replayed, this interferes with the timing between<BR>
digits and the actual digits themselves.<BR>
<BR>
The solution is to switch this to H323 (or SIP) and on your outbound dial<BR>
peer omit any type of dtmf-relay. This allows the g711 stream to carry the<BR>
dtmf natively to the headend.<BR>
<BR>
-nick<BR>
<BR>
On Thu, Mar 24, 2011 at 2:40 PM, Lelio Fulgenzi <lelio@uoguelph.ca> wrote:<BR>
<BR>
> That was my suspicion. Just wondering if those who went from SCCP to<BR>
> MGCP/H323 were encountering the same thing.<BR>
><BR>
> We'll have to see what the alarm vendor says.<BR>
><BR>
> I'm hoping someone on the list who has sccp/h323 configured ports on a<BR>
> vg224 can help with my other questions.<BR>
><BR>
><BR>
> ---<BR>
> Lelio Fulgenzi, B.A.<BR>
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<BR>
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<BR>
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>
> Cooking with unix is easy. You just sed it and forget it.<BR>
> - LFJ (with apologies to Mr. Popeil)<BR>
><BR>
><BR>
> ------------------------------<BR>
> *From: *"Jason Fuermann" <jason.f@shsu.edu><BR>
> *To: *"Lelio Fulgenzi" <lelio@uoguelph.ca>, cisco-voip@puck.nether.net<BR>
> *Sent: *Thursday, March 24, 2011 12:46:34 PM<BR>
> *Subject: *RE: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
><BR>
><BR>
> The head end is probably hanging up while the dialer is still spitting out<BR>
> codes, and the gw thinks it?s trying to dial an extension. Just a guess.<BR>
><BR>
> <A HREF="http://markmail.org/message/kwzlalbdo2asbxqo">http://markmail.org/message/kwzlalbdo2asbxqo</A><BR>
><BR>
><BR>
><BR>
><BR>
><BR>
> *From:* cisco-voip-bounces@puck.nether.net [mailto:<BR>
> cisco-voip-bounces@puck.nether.net] *On Behalf Of *Lelio Fulgenzi<BR>
> *Sent:* Thursday, March 24, 2011 10:55 AM<BR>
> *To:* cisco-voip@puck.nether.net<BR>
> *Subject:* [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
><BR>
><BR>
><BR>
> We have a remote site which uses alarms on their VG224. It seems they're<BR>
> not working properly. I've got a call back into the client to arrange a<BR>
> conference call with their alarm provider so I have access to technical<BR>
> details I can share.<BR>
><BR>
> When I look at the CDRs, I see some strange entries. Basically a call to a<BR>
> toll free number, followed by short calls of 1 second with invalid DNs. (see<BR>
> below)<BR>
><BR>
> I recall some talk about alarms and SCCP and it's better with MGCP. Can<BR>
> anyone who had this problem confirm if this was the same issue?<BR>
><BR>
> I have no problem setting up a few ports for MGCP but I have some concerns:<BR>
><BR>
> - can I have a VG224 with some ports as SCCP and some as MGCP?<BR>
> - how can I ensure that my MGCP ports work during failover, i.e. SRST?<BR>
><BR>
><BR>
> - I don't want to have to redo all my dial-peers. I don't mind putting<BR>
> in one dial that sends all calls from the VG224 to the SRST router, but I'd<BR>
> like an inbound dialpeer from the VG224 that assigns the same class of<BR>
> service for all inbound calls from that vg224.<BR>
> - Is this doable?<BR>
><BR>
><BR>
> ------------------------------<BR>
><BR>
> call 1<BR>
><BR>
> call 2<BR>
><BR>
> DateTimeOrigination<BR>
><BR>
> DateTimeOrigination<BR>
><BR>
> Mar 8, 2011 10:02:10 AM<BR>
><BR>
> Mar 8, 2011 10:02:34 AM<BR>
><BR>
> OriginalCalledPartyNumber<BR>
><BR>
> OriginalCalledPartyNumber<BR>
><BR>
> 1866------- <snip><BR>
><BR>
> 7000C<BR>
><BR>
> FinalCalledPartyNumber<BR>
><BR>
> FinalCalledPartyNumber<BR>
><BR>
> 1866------- <snip><BR>
><BR>
> 7000C<BR>
><BR>
> DateTimeConnect<BR>
><BR>
> DateTimeConnect<BR>
><BR>
> Mar 8, 2011 10:02:22 AM<BR>
><BR>
> N/A<BR>
><BR>
> DateTimeDisconnect<BR>
><BR>
> DateTimeDisconnect<BR>
><BR>
> Mar 8, 2011 10:02:34 AM<BR>
><BR>
> Mar 8, 2011 10:02:36 AM<BR>
><BR>
> Duration<BR>
><BR>
> Duration<BR>
><BR>
> 12<BR>
><BR>
> 0<BR>
> ------------------------------<BR>
><BR>
><BR>
> ---<BR>
> Lelio Fulgenzi, B.A.<BR>
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<BR>
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<BR>
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>
> Cooking with unix is easy. You just sed it and forget it.<BR>
> - LFJ (with apologies to Mr. Popeil)<BR>
><BR>
><BR>
> _______________________________________________<BR>
> cisco-voip mailing list<BR>
> cisco-voip@puck.nether.net<BR>
> <A HREF="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</A><BR>
><BR>
><BR>
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<BR>
------------------------------<BR>
<BR>
Message: 10<BR>
Date: Thu, 24 Mar 2011 18:48:13 -0400 (EDT)<BR>
From: Lelio Fulgenzi <lelio@uoguelph.ca><BR>
To: Nick Matthews <matthnick@gmail.com><BR>
Cc: cisco-voip@puck.nether.net<BR>
Subject: Re: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
Message-ID:<BR>
<1701739144.2856138.1301006893676.JavaMail.root@simcoe.cs.uoguelph.ca><BR>
Content-Type: text/plain; charset="utf-8"<BR>
<BR>
Thanks Nick. After talking with the client, it seems it's only one of a number of alarm panels that is causing the problem. So I'm not sure what the difference is between this unit and others. I'm hoping the client can set up a meeting with the vendor as soon as possible.<BR>
<BR>
Reading one of the threads, it talks about 4+2 or 4x2 as using touch tones and the contact ID using hook flashes. I wonder if switching modes will help.<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
---<BR>
Lelio Fulgenzi, B.A.<BR>
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<BR>
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<BR>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>
Cooking with unix is easy. You just sed it and forget it.<BR>
- LFJ (with apologies to Mr. Popeil)<BR>
<BR>
<BR>
----- Original Message -----<BR>
From: "Nick Matthews" <matthnick@gmail.com><BR>
To: "Lelio Fulgenzi" <lelio@uoguelph.ca><BR>
Cc: "Jason Fuermann" <jason.f@shsu.edu>, cisco-voip@puck.nether.net<BR>
Sent: Thursday, March 24, 2011 6:08:41 PM<BR>
Subject: Re: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
<BR>
Can almost tell you the problem from the title of the email:<BR>
<BR>
The problem with using MGCP/SCCP on these ports is that they are going to do DTMF relay. They will take the TDM tone, convert it to sccp, and replay it on the other side. This doesn't jive with alarm panels because their timing is very specific. They basically use DTMF tones like modem tones and send information back and forth via DTMF. Because SCCP DTMF messages have a fixed duration when they are replayed, this interferes with the timing between digits and the actual digits themselves.<BR>
<BR>
The solution is to switch this to H323 (or SIP) and on your outbound dial peer omit any type of dtmf-relay. This allows the g711 stream to carry the dtmf natively to the headend.<BR>
<BR>
-nick<BR>
<BR>
<BR>
On Thu, Mar 24, 2011 at 2:40 PM, Lelio Fulgenzi < lelio@uoguelph.ca > wrote:<BR>
<BR>
<BR>
<BR>
<BR>
That was my suspicion. Just wondering if those who went from SCCP to MGCP/H323 were encountering the same thing.<BR>
<BR>
We'll have to see what the alarm vendor says.<BR>
<BR>
I'm hoping someone on the list who has sccp/h323 configured ports on a vg224 can help with my other questions.<BR>
<BR>
<BR>
---<BR>
Lelio Fulgenzi, B.A.<BR>
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<BR>
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<BR>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>
Cooking with unix is easy. You just sed it and forget it.<BR>
- LFJ (with apologies to Mr. Popeil)<BR>
<BR>
<BR>
<BR>
From: "Jason Fuermann" < jason.f@shsu.edu ><BR>
To: "Lelio Fulgenzi" < lelio@uoguelph.ca >, cisco-voip@puck.nether.net<BR>
Sent: Thursday, March 24, 2011 12:46:34 PM<BR>
Subject: RE: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
The head end is probably hanging up while the dialer is still spitting out codes, and the gw thinks it?s trying to dial an extension. Just a guess.<BR>
<BR>
<A HREF="http://markmail.org/message/kwzlalbdo2asbxqo">http://markmail.org/message/kwzlalbdo2asbxqo</A><BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
From: cisco-voip-bounces@puck.nether.net [mailto: cisco-voip-bounces@puck.nether.net ] On Behalf Of Lelio Fulgenzi<BR>
Sent: Thursday, March 24, 2011 10:55 AM<BR>
To: cisco-voip@puck.nether.net<BR>
Subject: [cisco-voip] alarm panel issues and VG224 w/ SCCP<BR>
<BR>
<BR>
<BR>
<BR>
We have a remote site which uses alarms on their VG224. It seems they're not working properly. I've got a call back into the client to arrange a conference call with their alarm provider so I have access to technical details I can share.<BR>
<BR>
When I look at the CDRs, I see some strange entries. Basically a call to a toll free number, followed by short calls of 1 second with invalid DNs. (see below)<BR>
<BR>
I recall some talk about alarms and SCCP and it's better with MGCP. Can anyone who had this problem confirm if this was the same issue?<BR>
<BR>
I have no problem setting up a few ports for MGCP but I have some concerns:<BR>
<BR>
? can I have a VG224 with some ports as SCCP and some as MGCP?<BR>
? how can I ensure that my MGCP ports work during failover, i.e. SRST?<BR>
<BR>
<BR>
<BR>
<BR>
? I don't want to have to redo all my dial-peers. I don't mind putting in one dial that sends all calls from the VG224 to the SRST router, but I'd like an inbound dialpeer from the VG224 that assigns the same class of service for all inbound calls from that vg224.<BR>
? Is this doable?<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
call 1 <BR>
<BR>
call 2<BR>
<BR>
<BR>
DateTimeOrigination <BR>
<BR>
DateTimeOrigination<BR>
<BR>
<BR>
Mar 8, 2011 10:02:10 AM <BR>
<BR>
Mar 8, 2011 10:02:34 AM<BR>
<BR>
<BR>
OriginalCalledPartyNumber <BR>
<BR>
OriginalCalledPartyNumber<BR>
<BR>
<BR>
1866------- <snip> <BR>
<BR>
7000C<BR>
<BR>
<BR>
FinalCalledPartyNumber <BR>
<BR>
FinalCalledPartyNumber<BR>
<BR>
<BR>
1866------- <snip> <BR>
<BR>
7000C<BR>
<BR>
<BR>
DateTimeConnect <BR>
<BR>
DateTimeConnect<BR>
<BR>
<BR>
Mar 8, 2011 10:02:22 AM <BR>
<BR>
N/A<BR>
<BR>
<BR>
DateTimeDisconnect <BR>
<BR>
DateTimeDisconnect<BR>
<BR>
<BR>
Mar 8, 2011 10:02:34 AM <BR>
<BR>
Mar 8, 2011 10:02:36 AM<BR>
<BR>
<BR>
Duration <BR>
<BR>
Duration<BR>
<BR>
<BR>
12 <BR>
<BR>
0<BR>
<BR>
<BR>
<BR>
<BR>
---<BR>
Lelio Fulgenzi, B.A.<BR>
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<BR>
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<BR>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>
Cooking with unix is easy. You just sed it and forget it.<BR>
- LFJ (with apologies to Mr. Popeil)<BR>
<BR>
<BR>
_______________________________________________<BR>
cisco-voip mailing list<BR>
cisco-voip@puck.nether.net<BR>
<A HREF="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</A><BR>
<BR>
<BR>
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Message: 11<BR>
Date: Thu, 24 Mar 2011 22:24:46 -0700 (PDT)<BR>
From: Cisco Voip <cisco_newbie@yahoo.com><BR>
To: cisco-voip@puck.nether.net<BR>
Subject: [cisco-voip] SIP trunk configuration !<BR>
Message-ID: <614467.87105.qm@web113904.mail.gq1.yahoo.com><BR>
Content-Type: text/plain; charset="us-ascii"<BR>
<BR>
Dear all,<BR>
<BR>
I am sorry if its a sumb question. Can someone refer me any doc in which ITSP<BR>
side configuration is given for sip trunking solution on Cisco router.<BR>
<BR>
thanks<BR>
<BR>
<BR>
<BR>
<BR>
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Message: 12<BR>
Date: Thu, 24 Mar 2011 23:54:53 -0700 (PDT)<BR>
From: Paul <asobihoudai@yahoo.com><BR>
To: Cisco Voip <cisco_newbie@yahoo.com>, cisco-voip@puck.nether.net<BR>
Subject: Re: [cisco-voip] SIP trunk configuration !<BR>
Message-ID: <806089.67819.qm@web111313.mail.gq1.yahoo.com><BR>
Content-Type: text/plain; charset=us-ascii<BR>
<BR>
try this<BR>
<A HREF="http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html">http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html</A><BR>
<BR>
<BR>
<BR>
<BR>
<BR>
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<BR>
Message: 13<BR>
Date: Fri, 25 Mar 2011 03:06:32 -0700 (PDT)<BR>
From: Cisco Voip <cisco_newbie@yahoo.com><BR>
To: Paul <asobihoudai@yahoo.com>, cisco-voip@puck.nether.net<BR>
Subject: Re: [cisco-voip] SIP trunk configuration !<BR>
Message-ID: <740619.78202.qm@web113909.mail.gq1.yahoo.com><BR>
Content-Type: text/plain; charset="us-ascii"<BR>
<BR>
Dear Sir,<BR>
<BR>
Those are mainly case studies i think, where i can find a working example of SIP<BR>
trunking (ITSP side)<BR>
<BR>
Kindly help me<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
________________________________<BR>
From: Paul <asobihoudai@yahoo.com><BR>
To: Cisco Voip <cisco_newbie@yahoo.com>; cisco-voip@puck.nether.net<BR>
Sent: Fri, March 25, 2011 11:54:53 AM<BR>
Subject: Re: [cisco-voip] SIP trunk configuration !<BR>
<BR>
try this<BR>
<A HREF="http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html">http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html</A><BR>
<BR>
<BR>
<BR>
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Message: 14<BR>
Date: Fri, 25 Mar 2011 11:43:32 -0300<BR>
From: "Michel L. M. B. Perez" <michelmbperez@gmail.com><BR>
To: Cisco Voip <cisco_newbie@yahoo.com><BR>
Cc: cisco-voip@puck.nether.net<BR>
Subject: Re: [cisco-voip] SIP trunk configuration !<BR>
Message-ID:<BR>
<AANLkTimdaBgAnezfC2vDf9tBUJe3O2mrKLjkHjrUp0ZA@mail.gmail.com><BR>
Content-Type: text/plain; charset="iso-8859-1"<BR>
<BR>
Hello,<BR>
<BR>
This can be usefull. -><BR>
<A HREF="http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns832/822195_2.pdf">http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns832/822195_2.pdf</A><BR>
and<BR>
<A HREF="http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns833/911277.pdf">http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns833/911277.pdf</A><BR>
<BR>
Bye.<BR>
<BR>
============<BR>
Michel Perez<BR>
(48) 8409-0110<BR>
<BR>
<BR>
<BR>
2011/3/25 Cisco Voip <cisco_newbie@yahoo.com><BR>
<BR>
> Dear Sir,<BR>
><BR>
> Those are mainly case studies i think, where i can find a working example<BR>
> of SIP trunking (ITSP side)<BR>
><BR>
> Kindly help me<BR>
><BR>
><BR>
> ------------------------------<BR>
> *From:* Paul <asobihoudai@yahoo.com><BR>
> *To:* Cisco Voip <cisco_newbie@yahoo.com>; cisco-voip@puck.nether.net<BR>
> *Sent:* Fri, March 25, 2011 11:54:53 AM<BR>
> *Subject:* Re: [cisco-voip] SIP trunk configuration !<BR>
><BR>
> try this<BR>
><BR>
> <A HREF="http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html">http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html</A><BR>
><BR>
><BR>
><BR>
><BR>
><BR>
> _______________________________________________<BR>
> cisco-voip mailing list<BR>
> cisco-voip@puck.nether.net<BR>
> <A HREF="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</A><BR>
><BR>
><BR>
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Message: 15<BR>
Date: Fri, 25 Mar 2011 08:24:15 -0700<BR>
From: Mark Holloway <mh@markholloway.com><BR>
To: "Michel L. M. B. Perez" <michelmbperez@gmail.com><BR>
Cc: cisco-voip@puck.nether.net<BR>
Subject: Re: [cisco-voip] SIP trunk configuration !<BR>
Message-ID: <9CFE4E29-A75A-4AD1-8E1F-B3109EB1BD2E@markholloway.com><BR>
Content-Type: text/plain; charset="us-ascii"<BR>
<BR>
This is what you're looking for. Scroll down to Cisco Unified Border Element to SIP Service Provider and note all the various PDF's for different ITSP's.<BR>
<BR>
<A HREF="http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html">http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html</A><BR>
<BR>
<BR>
<BR>
On Mar 25, 2011, at 7:43 AM, Michel L. M. B. Perez wrote:<BR>
<BR>
> Hello,<BR>
><BR>
> This can be usefull. -> <A HREF="http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns832/822195_2.pdf">http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns832/822195_2.pdf</A> and <A HREF="http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns833/911277.pdf">http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns833/911277.pdf</A><BR>
><BR>
> Bye.<BR>
><BR>
> ============<BR>
> Michel Perez<BR>
> (48) 8409-0110<BR>
><BR>
><BR>
><BR>
> 2011/3/25 Cisco Voip <cisco_newbie@yahoo.com><BR>
> Dear Sir,<BR>
><BR>
> Those are mainly case studies i think, where i can find a working example of SIP trunking (ITSP side)<BR>
><BR>
> Kindly help me<BR>
><BR>
><BR>
> From: Paul <asobihoudai@yahoo.com><BR>
> To: Cisco Voip <cisco_newbie@yahoo.com>; cisco-voip@puck.nether.net<BR>
> Sent: Fri, March 25, 2011 11:54:53 AM<BR>
> Subject: Re: [cisco-voip] SIP trunk configuration !<BR>
><BR>
> try this<BR>
> <A HREF="http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html">http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html</A><BR>
><BR>
><BR>
> <BR>
><BR>
><BR>
> _______________________________________________<BR>
> cisco-voip mailing list<BR>
> cisco-voip@puck.nether.net<BR>
> <A HREF="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</A><BR>
><BR>
><BR>
> _______________________________________________<BR>
> cisco-voip mailing list<BR>
> cisco-voip@puck.nether.net<BR>
> <A HREF="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</A><BR>
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End of cisco-voip Digest, Vol 89, Issue 25<BR>
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