SIP Trunking between 2 different networks
Hi,
Can anyone confirms me that in order to establish SIP trunk between two IP PBX resided in different networks, what is the requirement?
providing that IP PBX doesn't have WAN port.
Hello omerpal1190,
You might want to look at Cisco Unified Border Element solution that will help you do SIP trunking easier. Kindly refer to the link attached for more information about the additional requirements that you might still need.http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-border-element/data-sheet-c78-729692.html
Hope this helps! If you need additional assistance regarding this, please feel free to send me an email at [email protected]
Regards,
Alyza Reyes
Similar Messages
-
How to create multiple sip trunks between cucm and cisco unified sip proxy
Dear Expert,
Is there a way to create multiple sip trunks between CUCM and Cisco Unified SIP Proxy (CUSP)? How to achieve it without creating multiple IP interfaces on the CUSP module.
CUCM: 8.5.1.10000-9
CUSP: 8.5.2
Thank you,
.wanHello Michael,
This SIP trunk is part of UCCE solution, which used between CVP, CUSP, and CUCM.
The requirements:
1) To have different codecs for different type of calls, as the phones are at few countries
2) To pass different number of digits from CUSP to CUCM for different call treatments
.wan -
Sip trunk between CUCM7.0 and third party VOIP provider
Hi all,
I'm looking for a solution/howto configuration for setting up a SIP trunk between CUCM7.0 and a SIP-VoIP provider.
Got SIP username, password and SIP-proxy IP from the provider.
I've done such a setup on CUCME a couple of times, but never on the CUCM.
Who can put me on right way?
Can it be done on the CUCM, or must an IOS-Device be used (got a PSTN-GW connected through H323 with CUCM)?
THanks for the hint,
Greets NorbertHere we go.....
CONFIG (Version=7.1)
=====================
Version 7.1
Cisco Unified Communications Manager Express
! Calling nr. incoming
voice translation-rule 40
rule 1 /\(.*\)/ /0\1/
! Discard prefix (calling nr.)
voice translation-rule 190
rule 1 /^0\(.*\)/ /\1/
rule 2 /^9\(.*\)/ /\1/
! Mapping, internat to external nr.
voice translation-rule 191
rule 10 /^[1-9].*/ /xxxxEXTERNALxxxx/
! for call-forwarding
rule 15 /^0\(.*\)/ /\1/
! Mapping external to internal nr.
voice translation-rule 192
rule 2 /^xxxxxEXTERNALxxxx/ /4xx/
voice translation-profile TP_IN_SIP
translate calling 40
translate called 192
voice translation-profile TP_OUT_SIP
translate calling 191
translate called 190
dial-peer voice 2001 voip
corlist outgoing dialCORnoFax
description *** SIP-TRUNK (OUT) ***
translation-profile incoming TP_IN_SIP
translation-profile outgoing TP_OUT_SIP
max-conn 2
destination-pattern 9.T
session protocol sipv2
session target ipv4:2xx.xxx.xxx.xxx
session transport udp
! customer external nr. range (one dot at the and -> 0-9)
incoming called-number xxxxxxxx.
dtmf-relay rtp-nte
codec g711alaw
no vad
gateway
timer receive-rtp 1200
sip-ua
keepalive target ipv4:2xx.xxx.xxx.xxx
authentication username xxEXTERNAL NR.xxxxx password 7 111111111111111111111
calling-info pstn-to-sip from number set xxEXTERNAL NR.xxxxx
retry invite 2
retry response 2
retry bye 2
retry register 2
retry options 1
registrar ipv4:2xx.xxx.xxx.xxx expires 60
host-registrar
Greets,
Norbert
Hope this help......Please rate if helpful -
SIP trunking between Microsoft OCS server and Cisco Voice GW router.
Hello All,
I have a client with an existing Microsoft OCS (office communications server) environment with the OCS server in their head office. The OCS clients in the remote Office registers with the OCS server in the head office. The WAN connectivity between the remote office and the Head office is MPLS.I would like to facilitate local call (PSTN) features at the remote site through a newly proposed Voice gateway router.
Can I achieve this by doing a SIP trunk between the OCS server in the head office to the newly proposed voice GW router in the remote office through the existing MPLS link. If yes, Could any one please assist me in this regards or suggest any other best solution to achieve the same.
Thank you in advance,
Mohammed Ameen RHi David,
this is a normal behaviour. To CUCM, OCS is a remote destination (just like your mobile phone). When your mobile phone hangs up, the system will put the call on hold for 10 sec.
This is there for the mobile user to go to his desk to pick up the call and continue the conversation (part of single number reach feature)
The best practise will be for the user to ensure that the other party hangs up the call first before he hang up.
Please grade if you think it's useful =) -
Can I share files between two different networks at same location?
Until TWC can offer DOCSIS 3.0 later this year or next...
- In my home I have a closet where all my audio, video and network come together.
- I have two cable modems from Time Warner
- Modem A goes to a WRT320N; 192.168.0.1
-- This router connects all my "heavy" downloads, ie. DirecTV receivers, Netflix, home server w/videos, gaming, etc.
- Modem B goes to a E2000; 192.168.1.1
-- This router connects my everyday use computers so they don't suffer poor speeds from all the traffic on Modem A
My question is, how or can I somehow share files between these two routers without the need for hitting the modems? Obviously I can just push files over the Internet but I am doing it at a much slower speed due to upload restrictions on the modem.
Thank you for any help on this!The easiest way would be to assign LAN IP 192.168.1.2 to the WRT320N and disable the DHCP server. Then assign static IP addresses in 192.168.1.* to all devices connected to the WRT at the moment, e.g. 192.168.1.10, 255.255.255.0, gateway 192.168.1.2, DNS 192.168.1.2 (or DNS servers of your ISP).
Now you can connect a LAN port of the WRT to a LAN port of the E2000. All DHCP clients get the IP address from the E2000 which will assign the E2000 as gateway, i.e. they use the E2000 for internet. All other devices will have static IP addresses and use the WRT as gateway. -
Problem with sip trunk between CCM and Huawei through Cisco ASA5520
Hello,
I have a next problem
During SIP conversation ASA is changing the ip address of CCM to corresponding name in ASA configuration inside the SIP packet:
To: <sip:443230282@Server_CCM1;user=phone>
ASA name configuration:
name x.y.z.h Server_CCM1
But it should be without any changes like that: To: <sip:[email protected];user=phone>. Because of that session cant be established. Remote SIP peer gives an error "Bad Request - 'Malformed/Missing URL"
When name was deleted in ASA "no name x.y.z.h Server_CCM1" we have no any problem with SIP initialization and call proccesing.
We are going to upgrade ASA from 8.2 to 8.3 and it seems that we will have the same problem because object will be created automaticly in new version (we are using a NAT) and we will not be able to delete an object like we did in version 8.2.
What configuration in ASA version 8.3 should be done to avoid this issue.
P.S Detailed debug from Huawei in attachment.
Thank you.Hi.
depending on your config, you might be hitting CSCta16361, this is fixed in 8.2(4)
if you can confirm it's still happening in latest 8.2 release, then a TAC case needs to be opened so investigation is done and a new bug is opened.
if you've tested 8.2(4) already and it's still doing the same, then a TAC Service Request should be opened for more investigation and possibly opening a new defect.
Best regards,
Fadi.
does this answer your question? if yes please mark it resolved. -
Dear All,
We have two cucm Clusters in Different Locations between that clusters i created
Inter-Cluster Trunk (Non-Gatekeeper Controlled) Now all are working fine Between Clusters
audio calls & Video calls between sccp 8945 phones , but iam facing a Problem with third party
Video Phones (Polycom VVX 1500 ) Third Party SIP Phones located in second cluster, From 1 st cluster cisco 8945 Video
phone to 2nd cluster Polycom Video phone all calls are works for voice call only, but no video ,
Please Suggest me Solution.
Thank you,
SrimanTry setting up a SIP trunk between the two clusters and set a route patten just to the VVX 1500 and check how that goes.
From memory inter-cluster trunks are a H.323 like protocol which might have video inter-op issues with the Polycom device. -
Why do we need MTP in the SIP trunk for CVP warm transfers
Hi All,
Why do we need to enable MTP in SIP trunk between CUCM and CVP for CVP based trasnfers???
Thanks in advance!!
Regards,
Thammaya Gupta K.I saw also in the CDR logs that the IP Phone media transport going to CUBE is in G711.And as well in the wireshark capture of the IP communicator that the CUCM invoke to use the g711 codec but as per ITSP logs they are now in the g729.
@ Jamie If I un-tick the MTP point required in SIP trunk will make the call leg from IP Phone to CUBE g729 (w/o hw resource), I have also tried to use g729 preferred originating codec, but still the IP Phone is using g711.
I have seen a documentation states:
" To configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or transcoder that supports the G.729 codec." - I read this on the CUCM help page under configuring SIP trunk setting.
Our ultimate goal is to use g729 without using HW MTP/ transcoder.
IP Phone ->CUCM SIP Trunk ->CUBE-> ITSP -
Hello,
We are in stage of configuring SIP Trunk between 2 CUCM clusters (cluster A & cluster B ) via SME.
Can any body help how to configure incoming & outgoing calls on SME using SIP ?
For Example:
For a call from user A (cluster A) to user B (cluster B)
User A from Cluster A will use Route Pattern for outgoing calls, what parameters need to be configure in SME to answer call coming from cluster A ?
Then for SME outgoing call towards cluster B, do we need to configure RP ? In cluster B what parameters need to be configure to answer call coming from SME ?
I am trying to understand the call leg step by step from one CUCM cluster to other through SME.
We have 3 Digits of extension in Cluster A & we have 4 Digits of extension in Cluster B.
Full numbers of ip phone in each cluster is 8 Digit & we want 8 digit calling pattern to call from cluster A to B & vice versa.
Thank in advance.
Regards,
ConieGot it.
We have only one SME server at time & we are planning to use Inter-cluster trunk between clusters as backup incase if SME stop working for any malfunctioning reason.
For inter-cluster trunks we will have :
1- Inter-Cluster trunk in Cluster A point to Cluster B ?
2- Inter-Cluster trunk in Cluster B point to Cluster A ?
3- In cluster A RL, set inter-cluster trunk for cluster B as secondary ?
4- In cluster B RL, set inter-cluster trunk for cluster A as secondary ?
should i use same calling approach as we discussed above via Route Patterns only ?
As we have CUCM 8.6 in both clusters, we are planning to use H.323 Inter-Cluster (Non Gatekeeper) Trunk as backup.
And PSTN lines as backup for H.323 Inter-cluster.
Do you have any comments regarding our design ? are we on right track with this failover approach ?
Regards,
Conie
Thank you to being answering my questions appreciated your answers helped me alot. -
CUCM - trunking between 7 and 9
Hi,
Document needed to do a trunking between cucm 7 and cucm 9 ( intercluster trunking)
We are planning to route a PSTN call from IP phones which are registerd with old CUCM on version 7 from our new cucm 9 which has FXO- line active.
how do I achive it.
RegardsHello
You have to use intercluster non-gatekeeper controlled . I do nothink there is special document for this , but you can see the below link:-
https://supportforums.cisco.com/discussion/11766291/intercluster-sip-trunk-between-cucm-version-71-and-91
Thanks
please rate all useful ifnromation -
How to Integrate Microsoft Lync 2010, Asterisk, and a sip trunk.
Dear Friends.
i need you to assist me to step my new project
Objective:
Setup Asterisk
to Configure a SIP trunk between Asterisk and the SIP provider of my choice
Integrate Lync Server 2010 with Asterisk
Configure a dial plan
Configuring Voice Polices, PSTN Usage Records, and Voice Routes.
To be able to make international
local call to any mobile extension or same number range
This is a new project to me can anyone please simply assist me step by step ?
Thanks
GreenmanHi GreeMann, Which Flavor of Asterisk you are using ex: FreePBX, Elastix, AsteriskNow.
You can use any of them most of the configuration will be similar.
To configure the SIP Trunk of service provider in asterisk check this
http://wiki.freepbx.org/display/ST/Setting+up+SIPStation+manually+in+FreePBX http://wiki.freepbx.org/display/F2/Trunk+Sample+Configurations
Here is my blog Step by step guide to Integrate asterisk ( Elastix) with Lync
http://mslyncforall.blogspot.in/2014/12/lync-2013-asterisk-pbx-integration.html
http://blogs.technet.com/b/rischwen/archive/2013/08/21/series-exchange-2013-and-lync-2013-integration-with-asterisknow-pbx-pt-1.aspx
Please let me know if you encounter any issues i am happy to help you.
Whenever you see a helpful reply, click on Vote As Helpful & click on Mark As Answer if a post answers your question. -
Telco Messages via PRI (VG) connected to CUCM 9.X via SIP Trunk??
Hello
Questions:
Should I hear tel-co message on an IP Phone if a call that is meant to be long distance is sent to the gateway without a one (1). Currently these calls simply ring until disconnect, but work properly if the user dials a 1, the user expects to get a message from the provider and I am wondering if the SIP trunk between GW and CUCM is not allowing it?
Scenario:
IP Phone --> CUCM (SIP Trunk) --> ISR 2901(PRI) --> PSTN
In CUCM we have a local pattern 9.[2-9]XX[2-9]XXXXXX
If the IP Phone dials a number that is actually a long distance number, which would require a 1, but they dial it as a 10 digit local number, the call gets to the gateway and IP Phone hears ringing but it, rings until it eventually disconnects. At this point I believe the gateway is sending a cause code back to CUCM and I would expect error message back from the telco informing the caller of the issue, such as "Please dial 1 before a long distance number". Is there a specific setting on the SIP trunk and or Gateway to achieve this?
See Q931 Debug
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838F
Exclusive, Channel 15
Display i = 0xB1, 'London Hydro'
Calling Party Number
UCS5-GW-02# i = 0x2181, '5193334444'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '5191112222'
Plan:ISDN, Type:National
*Jul 23 15:33:28.363: ISDN Se0/0/1:23 Q931: RX <- STATUS pd = 8 callref = 0xC3FA
Cause i = 0x80AB28 - Access information discarded
Call State i = 0x01
*Jul 23 15:33:28.411: ISDN Se0/0/1:23 Q931: RX <- CALL_PROC pd = 8 callref = 0xC3FA
Channel ID i = 0xA9838F
Exclusive, Channel 15
*Jul 23 15:33:28.415: ISDN Se0/0/1:23 Q931: RX <- PROGRESS pd = 8 callref = 0xC3FA
Cause i = 0x80FF - Interworking error; unspecified
Progress Ind i = 0x8088 - In-band info or appropriate now available
Thanks
RichardYes, Early media needs to be turned on the SIP Profile that is associated to the SIP trunk. The setting is SIP Rel 1xx options. It needs to be set for Send PRACK for all 1XX messages. If you have CUCM 7.x , this setting is a service parameter.
-
Hi Guys, we are going to setup a few SIP Trunks between CUCM 7.1.3 and third party FAX server. Do we need extra license for this SIP trunk? I had a look at the license calculator but it doesn't mention any SIP trunks. So I think it will not consume any license units, am I correct?
Thanks. LeoYes, correct, no licenses required/consumed for/by SIP Trunk.
GP.
Pls rate the post if it helps !! -
A SIP trunking connection that scales with growth
Since delivering its first inflight Internet experience in 2008, Gogo has grown to over 800 employees in five offices around the country. With 1000s of customers thirsty for in air data connections, its crucial that Gogo is highly available to support its staff and its airline partners globally. Capacity was near its max, but the […]
Read More
This topic first appeared in the Spiceworks CommunityHi,
No as you said you have already created the Trunk between the Sonus and Lync you can use the same trunk. but in the Sonus you have to configure the outbound routes from which Trunk you have to send the calls for the new Trunk provider or the Level3.
check this
https://support.sonus.net/display/UXDOC41/SIP+Trunking+Between+SIP+Border+Elements
Whenever you see a helpful reply, click on Vote As Helpful & click on Mark As Answer if a post answers your question. -
DTMF not working between 2 CUCM SIP Trunks
Dears
We have configured 2 SIP trunks on 2 CUCM servers , all calls are working fine except the DTMF ?? any ideas what can enable the DTMF between the SIP Trunks??
Inter-Cluster Trunk (Non-Gatekeeper Controlled) On Both Sides and the Codec is G.711u
Best RegardsOk, so as I understand following is the issue description.
There are 2 sites A & B. A has CUCM cluster for IPT users & Site B has CUCM Cluster for Contact Center users. These 2 clusters are connected using Non-GK controlled ICT.
Site A users when call Site B IVR, they hear the greeting but DTMF is not recognized hence they are unable to choose between options. Correct ?
1. What are the CUCM, IP IVR or UCCX/UCCE versions ?
2. Are the site B users able to choose options without any issues ?
3. When you said IVR, is it IPIVR/UCCX/CVP, Unity Connection, Unity, CUE or BACD on gateway ?
Please, always give full details of your setup & then ask the query, as it helps you only to get a quick & precise answers.
GP.
Pls rate helpful posts by clicking on stars below the post !!
Maybe you are looking for
-
Can I send an E-Mail with photo (not as an attachment)?
I would like to send an E-Mail with a small photo (not as an attachment) to someone who is reluctant to open attachments. Can I do this and how?? Thank you tch
-
GROUP BY - Is there a way to have some sort of for-each statement?
Hi there, This discussion is a branch from https://forums.oracle.com/thread/2614679 I data mart I created for a chain of theatres. The fact table contain information about ticket sales, and I have a some dimensions including DimClient and DimTime. He
-
Does anyone know how to group songs by album, and then by artist. Essentially I say this because of compilations. Generally I like to keep things sorted by artist. The only problem with this is the fact that I have several compilations, so sorting by
-
Error in Custom "Non Po Invoices" module
Hi All, I did migrate the oracle 11i custom "Non Po Invoices" module into R12. I'm able to run the "Create Non Po invoice page" without any error, but when I try to submit the invoice, I'm getting error like "Invoice Number - Set method for attribute
-
How long is the "preparing for dispatch" stage for the iphone 5s?
How long will my iphone 5s be in the "preparing for dispatch" stage? Thanks