Branch call PSTN routing

Client of HQ and Branch office with 512 line MPLS to only HQ they had E1 trunk, branch to call PSTN the call had to go via MPLS in HQ and the call RTC branch terminated at HQ only, is it a good design or better we need to add an ISDN line at the level of the branch to end the PSTN call.

in terms of good design, it is not

because this design has some imitations like below

1-single point of failure, if the MPLS link is near the bottom of the branch cannot be accessed via PSTN and they will lose access to the PSTN, too

2 - the MPLS link will be as a bottleneck for this branch because the PSTN and onnet calls will go on the PSTN connection

However if this is due to the customers needs/budget then you can do it like that, but don't forget to mention the recommendations for local PSTN

HTH

If useful rates

Tags: Cisco Support

Similar Questions

Configure incoming calls to route to the internal unit

I have a Cisco 2921 router which has a 4 FXO inside card. I would like to configure so that ALL incoming calls on all 4 ports to be forwarded to a post internal (1001), it is a test environment and I can't seem to understand what Miss me. The config is below:

Building configuration...

Current configuration: 8500 bytes

!

! Last configuration change at 08:19:46 EST Friday, March 1, 2013 by sjones

!

version 15.1

horodateurs service debug datetime msec localtime

Log service timestamps datetime msec localtime

no password encryption service

sequence numbers service

!

hostname WH-VOIP-2900

!

boot-start-marker

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logging buffered 10000000

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AAA authentication login default group Ganymede + local line

/NOAUTH AAA authentication login no

default AAA authorization exec group Ganymede + local no

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orders accounting AAA 15 by default start-stop Ganymede group.

Default connection accounting AAA power Ganymede group.

!

!

!

!

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AAA - the id of the joint session

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clock timezone IS - 5 0

summer time clock IS recurring

!

No ipv6 cef

IP source-route

IP cef

!

!

!

!

!

no ip domain search

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!

Authenticated MultiLink bundle-name Panel

!

!

!

!

!

!

FXO trunk group

!

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name of the object cn = IOS - Self - signed - certificate - 3979560690

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!

!

TP-self-signed-3979560690 crypto pki certificate chain

certificate self-signed 01

308201B 6 A0030201 02020101 3082024D 300 D 0609 2A 864886 F70D0101 04050030

2 060355 04031326 494F532D 53656 C 66 2 AND 536967 6E65642D 43657274 31312F30

69666963 33393739 35363036 6174652D 3930301E 170 3130 31323232 31333533

30375A 17 0D 323030 31303130 30303030 305A 3031 06035504 03132649 312F302D

4F532D53 5369676E 656C662D 43 65727469 66696361 74652 33 39373935 65642D

36303639 3030819F 300 D 0609 2A 864886 01050003, 818, 0030, 81890281 F70D0101

8100DD47 9227149F 2D084CE5 3 D 7DBF4FCA 227595 C3519000 3F468821 D56F653A

E74FCBAD B4936598 F0C26B2B 6132ADE7 1B1BDC89 44D3C53F 63DDAF78 8E08FCA7

7044095A DBE38889 7CD 48871 94ED1CF9 F2ECC50A 8BD21AFC 5BC3B3FC B322E161

F3CE339A 88AA803B E3705349 03A7D918 C11E5844 ECF039EB FEC44CDF 52A59AE5

0C 430203 010001A 3 75307330 1 130101 FF040530 030101FF 30200603 0F060355

551 1104 19301782 1557482D 564F4950 2 D 302E6D67 323930 73642E65 6475301F

23041830 16801463 9BA90049 2F6005DC F2A35FC3 0EDB2530 0603551D 4138 329D

1 D 060355 1D0E0416 0414639B A900492F 6005DCF2 A35FC332 9D41380E DB25300D

06092A 86 01010405 00038181 005C2C45 9F687AEF 3219F567 337E55CD 4886F70D

9E524A1B 7879B3B1 F3C872F9 DFF7F014 FFE0D84B 67252EFE 3DFF8959 9565ADE2

79857E34 FFF2C3DE 667D5D62 8A4E4690 D874CF4A 8B 180832 7748D1E8 BB71543B

BC404126 02DABACB DDF24EE6 6F63F8CE F7F8494C 66115C B768BC77 DA2D5C2C 77

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quit smoking

voice-card 0

DSP services dspfarm

!

!

!

voip phone service

h323 connections allow h323

allow connections h323 to SIP

allow connections sip h323

allow sip to sip connections

redirect ip2ip

Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none

H323

!

voice class codec 1

g711ulaw codec preference 1

codec preference 2 g729r8

!

vocal h323 class 1

H225 timeout tcp establish 3

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!

!

!

!

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license udi pid CISCO2921/K9 sn FTX1448AJ6B

HW-module pvdm 0/0

!

!

!

username admin privilege 15 secret 5 $1$ iKc / $uQJli0iQG9VAu4PiFeYC8 /.

!

redundancy

!

!

!

!

!

!

!

!

!

!

!

interface GigabitEthernet0/0

Description inside LAN

IP 10.40.0.51 255.255.0.0

automatic duplex

automatic speed

H323-gateway voip interface

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!

interface GigabitEthernet0/1

no ip address

Shutdown

automatic duplex

automatic speed

!

interface GigabitEthernet0/2

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automatic duplex

automatic speed

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IP forward-Protocol ND

!

IP http server

local IP http authentication

IP http secure server

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Server enable SNMP traps snmp authentication linkdown, linkup warmstart cold start

Enable SNMP-Server intercepts ATS

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Server enable SNMP traps insertion withdrawal flash

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RADIUS-server host 10.60.253.10 key Pa$ $word

RADIUS-server application made

!

!

control plan

!

!

voice-port 0/0/0

1 FXO-group of circuits

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!

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connection ÉRA 1001

!

voice-port 0/0/2

3 FXO trunk-group

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!

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!

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the associated profile 2 registry WH-2900_MTP

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Forward-digits 10

!

voice pots Dial-peer 5

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alert progress_ind activate 8

progress_ind enable progress 8

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alert progress_ind activate 8

progress_ind enable progress 8

prefix 011

!

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composition of 4-digit SRST Description to other sites

destination-model 2...

Forward-digits all the

prefix 704658

!

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progress_ind disconnect switch 8

session target ipv4:10.65.0.23

codec voice-class 1

h323 voice-class 1

DTMF-relay h245 alphanumeric

rate of 14400 Fax

IP qos dscp cs5 signaling

No vad

!

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codec voice-class 1

h323 voice-class 1

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rate of 14400 Fax

IP qos dscp cs5 signaling

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!

!

!

!

access controller

Shutdown

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!

Call-Manager-emergency

secondary-tone 9

MAX conferences 4-6 win

transfer full-consult system

3 timeouts interdigit

IP source address 10.40.0.51 port 2000

Max-joined 50

Max - dn 100 double line

primary phone message system is offline

secondary system message standalone

1 7046582 model numbering plan... extension-length 4

transfer-model. T

KeepAlive 10

voicemail 2525

call-Park select non-auto-match

ground of appeal forwards. T

call forward availability 97046582525

timeout before call 97046582525 16 noan

aa-mm-dd date format

!

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VM integration

direct model * GNC

peer-to-peer of nonresponse 5 FDN of mires * GNC *.

peer-to-peer busy 7 FDN of mires * GNC *.

safe-to-post non-response 4 FDN of mires * GNC *.

safe-to-position 6 FDN of mires * GNC *.

!

!

Line con 0

password V01pG8te

line to 0

line vty 0 4

access-class 23 in

privilege level 15

password V01pG8te

transport input telnet ssh

line vty 5 15

access-class 23 in

privilege level 15

password V01pG8te

transport input telnet ssh

line vty 16 1114

transport of entry all

!

Scheduler allocate 20000 1000

NTP 129.6.15.29 Server

end

Jeff,

I guess that 100 & 101 voip dial peers point to a CuCM?

The destination model on the voip dial peer does not 1001 on the ERA and they must change to something like: -.

Dial-peer voice voip 101

voice mail Dial 100

destination-model [12]...

voice mail Dial 101

destination-model [12]...

destination-model [2]...

Hope this helps,

Craig

PLEASE EVALUATE THE MESSAGES USEFUL

VCS - CPL to block call routing to and from the same area of neighbor

Hi all

Is there a CPL rule that can be written to block calls from a neighbouring area to the same area of neighbor?

We have problems with CUCM sent appeals to the our VCS VCS then routes the call return CUCM, causing thus "too many jumps."

In the example

MOVI user control VCS try to compose endpoint registered RTC CUCM, but incorrectly dials

Appeal does not have to search for the terms of local area

Call does not match search for VCS Expressway rules

Call corresponds to captures all of CUCM (. *) @domain.com

CUCM does not form DN or URI for the wrong phone number and the call is routed to the VCS control due to the SIP routing model of *. * on CUCM

We have to capture all CUCM at the end, since we offer PSTN, as well as our users and phones have the alphanumeric URIs assigned to the DNs for Jabber, etc.

I fear that he is not really an easy way to do it, I know because the call charged CPL first, CPL don't know what area it was intended for.

It's probably more easily accomomplished with a slight modification to your rules of research rather than spraying anthing "@domaine" everywhere - something like:

Anything incoming directly at @domaine in the nearby area of the local area (as is inside your organization, such as if the CUCM not found and sent it to the VCS he won't be on the CUCM, so must be local (or nonexistent) - and stop.

And the opposite of the above - route whatever it is for @domaine of the localZone.GetDaylightChanges (because it was not found locally) to stop and the CUCM neighbour-

And do similar for what is coming from other areas outside to route to wheverver that will be endpoints. In this way, your call should not be bouncing back between the CUCM and VCS and create the loop that you are currently experiencing.

Of course, if your environmetn is a little more complex than the simple CUCM and VCSes, this can be a bit oversimiplified, but could lead you in the right direction.

Another thing to consider too - if you put in your endpoints in a directory, users can use - that will help you to prevent wrong type stuff :)

Wayne
--
Remember the frequency responses and mark your question as answered as appropriate.

DX 80 - dial PSTN call does not

We have setup in the video below. But in recorded in CUCM 10.5 x SX80 try calling PSTN numbers, does not.

Hit gateways voice with error, error of unavailable resource. But other devices work properly with the same number of rank. Help, please

Hello

Can you put the codec g729r8 on your existing voice class codec and affect your pointing to your (not only G711ulaw) CUCM voip dial-Exchange.

Also configure transcoding on router resources and register with CUCM.

In my view, that it is somewhat related to DSP.

Kind regards

Raclot.

ISDN call routing

Hi all

What is the best way to configure the call routing suitable for ISDN under the following conditions:

-two gateways ISDN deployed, each at a different location,

-the two gateways to join prefixes with the same VCS,

-each gateway is in its own sub-area

-each group of endpoints assigned to subareas according to their location.

The goal is to have a defined range of endpoints in a specific location (subzone), calls are routed through the correct gateway.

Should this be done by the removal of specific links, by policy or any other method call?

And what about redundancy in such a deployment (if one of the bridge falling down)?

Kind regards

Maciek

First of all separate the user experience with what is done on a technical level.

(Registering a GW ISDN with a different prefix or no prefix at all) does not mean that

the user cannot reach it or that the user can dial a specific prefix.

Technology information is not only one way to go, there are often several ways which lead to the same goal.

More your deployment depends on the capabilities of your team, the requirements...

With X7.2 you can define a search based on a box named (void) rule and match your ISDN line prefix for ISDN gw1 and put other users in another area, rule of similar research with the same prefix that the user made up but this time

matched with the other isdngw.

For the ISDN gw you could either map via various registration prefixes or of what I prefer, put them

each in their own areas.

If you have these two rules search and add two more with a lower priority, it could

work for a failover as well, but I would use a CPL or NPS server for such a scenario which

you would better card on the case of the error.

The VCS is ideal because it is quite simple and logical, but still has CPL policies to extend a lot.

In any case it is not the best out of the box call control, so if you need that, look in CUCM.

Call the question of routing Manager

Allows a calling Manager route to an extension based on the time of day? For example, a call incoming during opening hours, calling press 0 and the call goes to the operator. Incoming calls during the closing hours, calling press 0 and the call will the night watchman. Is this possible?

Thank you - Pat

If you use Cisco Unity Connection, you can use the system lists to accomplish your question. Basically, you say: 0 option to send a Manager to call "Operator called" or something else. you attach a schedule to it and use the rules of transfer closed & STandard. During the transfer of the normal working hours (hourly) to X and during the transfer closed at X. Another way is to route the call to CUCM and using TOD routing in CUCM. The easiest way is to use the unit.

Endpoint VC PSTN call

Hello

I have a version of control VCS 7.2 integrate with 8.6 CUCM via SIP trunk.

The VC endpoint can call IP phones and vice versa.

I configured a search rule (9\d {8}) (9 + followed by 8 digits) in VCS to enable VC endpoint of RTC via CUCM appeal.

But it fails somehow.

When I checked the history of calls in VCS, seems that no rule is hit... I can't find that record everything that VC endpoint called PSTN.

You have an idea on that?

Thank you.

Try adding @. * at the end of your string pattern as your endpoints can be sent [RTC number]@yourdomain.com rather than just the numbers.

You see the research come via the history research and it of simply not to trigger a search rule, or do you see no research at all the?

Caller ID of VCS Jabber Video on SIP Trunk on the phone

Hello, Netpros.

I am trying to find a way to show a caller E.164 ID when a call to a video client Jabber a UCM or a PSTN phone.

VCSC/E 7.1; 13.2 TMS, configured TMSPE.

We have a VCSC > SIP Trunk > UCM (8,6) > H.323 GW w / PRI voice configured for the composition of and the PSTN destination. We do this, so the Jabber client video can be used as a softphone, and so we can have transparent FindMe to voice and SNR to the video in either sense. In TMS, we have a model of implementation configured with an address of the video, ILA and unit address. He pulls ad office phone number for video of Jabber users.

From what I read here:

http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_FindMe_Express_Deployment_Guide_X7-1.pdf, we should be able to see the caller ID when you call send out an ISDN GW.

«This section describes how to use FindMe with calls are routed through an ISDN gateway (for example, when you call a mobile phone, or any other destination accessible ISDN).» If VCS has Caller ID (Applications > FindMe > Configuration) defined to use the ID FindMe, the identification of the appellant presented will be the user E.164 phone number. »

UCM SIP config is vanilla, by

ttp: / /www.tandberg.com/collateral/documentation/Deployment_Guides/Cisco_VCS_Ci...

However, when a video call Jabber is directed to a University Complutense of MADRID registered phone, we get "unknown caller". When to call us directly to the PSTN, we see that the BTN from the PRI.

What is strange, is that if we call another client of Jabber of VCS (with all the same problems), directly into a phone UCM via Internet on URI (for example [email protected] / * /) through the VCSE, we see the caller ID name.

Thank you, all!

Joshua

Hi Joshua,.

At present the VCS will only send caller E.164 id to a GW of H.323 ISDN, which is registered for VCS.

There a feature request in the order book to define if E.164 Caller ID or name/Video address on a per-zone basis. This will also be sove the question when the call is to go on a SIP connection, as in this case, a remote ISDN GW.

Thank you

Guy

SPA9000 And direct Ip calls?

Hi, I just bought a SPA9000, and I wonder if I can set for direct ip calls.

I use the FXS 1 on the spa9000 and line 1 is already on my VOIP provider.

PBX settings
Call the routing rule: (4xx<:L1>. |) <:L2>8xx. | <:L3>7xx. | <:L4>6xx.)

And 1 line numbering Plan is:
(<><:@mydynds.com:5061>xxxxxxx)

Therefore, the call and the newspaper

-----------------------------------------------------------------------
[3] RegOK. NextReg in 58 (1)
2. number of report 1 (1)(40 ms)
Calling:[email protected]:6060
[0:0] DIAL the ALLOC AUD (port = 16434)
[0:0] RTP Rx upward
[JIU] CallRoute:L144549141
PRI--> INVITE--> pub
Calling:[email protected]:5061
Bcc:ringback / CallProgress 2
CC:ringback
[0:0] RTP Rx Dn
Pub - 200--> CCI - 200--> pri (appeal)
[0:0] ENC INIT 0
[0:0] RTP Tx Up (pt = 0-> c0a80165: 16438)
[0:0] RTCP Tx upward
----------------------------------------------------------------------
But after some time 20 seconds or maybe less she then drops the call

RespLooseMatch
[0:0] 16 (2) LAT -.
RespLooseMatch
RespLooseMatch
PRI says BCC GUEST: pri - 200--> CCI - 200--> pub
CC: finished
[0] alert FM stop RxTx (c = 002c67dc; a = 0)
[0:0] call of Rel AUD
359dc0 terminated DLG
Sess end 3668ec
TP Analyzer error: 34
[0] on the hook

-----------------------------------------------------------------------

Any ideas?

Thanks in advance.

I tried to research on your problem and I found another element which may help us: Lets guess 2 devices and b. also allows assume address IP is 192.168.1.50 and B device IP address 192.168.1.51.

Configure these settings in device A:

Step 1: Line 1 tab, set port SIP SIP settings section: 5060.
Stage 3: In line 1, Proxy and registration section all the empty fields or no. except to call without Reg: YES and years without Reg: YES.
Step 5: Tab of row 1, section of subscriber information full name value: user ID: 11 and leave the other fields swamp.
Step 7: In the tab section of plan Dial Dial Plan of line 1 on the Court together (21S 0<:@192.168.1.51:5060>), enable the IP numbering: YES

Configure these settings in device B:

Step 8: Line 1 tab, set port SIP SIP settings section: 5060.
Step 10: In line 1, Proxy and registration section all the empty fields or no. except to call without Reg: YES and years without Reg: YES.
Step 11: Tab of row 1, section of subscriber information full name value: user ID: 21 and leave the other fields swamp.
Step 13: In the tab section of plan Dial Dial Plan of line 1 on the Court together (11S 0<:@192.168.1.50:5060>), enable the IP numbering: YES

Now, how does it work:

-To call device 1 line of device (any line) B dial 11.
-To call device B line 1 of the device (one line) dial 21.

This has proved works with two SPA phones that are configured in the same local network segment. But in you I thing deal you make IP dial in on a remote network that would probably be another network. In this case, I think that you must configure the VPN between the network where the SPA-9000 and the other phone remotely. Then configure the IP numbering as as it is done in the above procedures.

I hope this helps.

SPA3102 will not dial a number, or forward calls to Elastix

For the past 2 days, I am trying to get this SPA 3102 to work as a PSTN gateway but no luck... My network is 172.16.1.0/24. My default gateway is 172.16.1.1 and local IP address of the SPA is 172.16.1.200. Elastix server is 172.16.1.8. I don't have any other VOIP providers so I na not configure my WAN for SPA settings (Nothing connected to the WAN port)

Here are the screenshots and my config settings:

The SIP Trunk settings:

Outbound caller ID: 383579 (my local PSTN number)

Chanels maximum: 1

Name of the Trunk: pstn

Parameters of peers:

do not allow = all
allow = ulaw
Polycom = no
context = of-RTC
dtmfmode = rfc2833
Host = 172.16.1.200
incominglimit = 1
NAT = never
port = 5061
qualify = yes
secret = 300
type = friend
username = pstn
Entrants settins empy rest under vacuum

Road of basketing:

DID number: 383579

Transfer the call to ext: 205

rest is default or is empty

Line1 SPA is disabled because I want just difficulty dailing and receive phone calls.

Here are a few screenshots of RTC in SPA line configuration:

As see you in picture above I have sound but spa isn't answer...

It comes from the phew journal asterisk call out:

-[358291@from-internal:1] Macro execution ("SIP/205-087af3d8","user-callerid |") SKIPTTL | ") in new stack
-Execution [s@macro-user-callerid:1] Set ("SIP/205-087af3d8", "AMPUSER = 205") in new stack
-Execution [s@macro-user-callerid:2] GotoIf ("SIP/205-087af3d8","0?") report") in new stack
-[S@macro-user-callerid:3] ExecIf execution ("SIP/205-087af3d8","1" | ") The value | REALCALLERIDNUM = 205 ") in new stack
-Execution [s@macro-user-callerid:4] Set ("SIP/205-087af3d8", "AMPUSER = 205") in new stack
-Execution [s@macro-user-callerid:5] Set ("SIP/205-087af3d8", "AMPUSERCIDNAME = Damir here") in new stack
-Execution [s@macro-user-callerid:6] GotoIf ("SIP/205-087af3d8","0?") report") in new stack
-Execution [s@macro-user-callerid:7] Set ("SIP/205-087af3d8", "AMPUSERCID = 205") in new stack
-Execution [s@macro-user-callerid:8] Set ("SIP/205-087af3d8", "CALLERID (all) ="Damir Reic ' <205>' ") in new stack
-Execution [s@macro-user-callerid:9] Set ("SIP/205-087af3d8", "REALCALLERIDNUM = 205") in new stack
-[S@macro-user-callerid:10] ExecIf execution ("SIP/205-087af3d8","0 |") The value | Channel (Language) = ") in new stack"
-Execution [s@macro-user-callerid:11] GotoIf ("SIP/205-087af3d8","1?") continue ") in new stack"
-Goto (macro-utilisateur-callerid, s, 20)
-Execution [s@macro-user-callerid:20] NoOp ("SIP/205-087af3d8","Using CallerID"Damir Reic" <205>" "") in new stack
-Execution [358291@from-internal:2] Set ("SIP/205-087af3d8", "_NODEST =") in new stack
-[358291@from-internal:3] Macro execution ("SIP/205-087af3d8","record-enable |") 205. OFF | ") in new stack
-Execution [s@macro-record-enable:1] GotoIf ("SIP/205-087af3d8","1?") check ") in new stack"
-Goto (macro-record-enable, s, 4)
-AGI [s@macro-record-enable:4] performance ("SIP/205-087af3d8","recordingcheck |") 20090827 205946 | 1251399586.64 ") in new stack"
-Launch Script AGI/var/lib/asterisk/acted-bin/recordingcheck
recordingcheck |-205946 20090827 | 1251399586.64: outgoing recording not enabled
-Recordingcheck AGI completed Script, return 0
-Execution [s@macro-record-enable:5] MacroExit ("SIP/205-087af3d8","" "") in new stack
-[358291@from-internal:4] Macro execution ("SIP/205-087af3d8","dialout-trunk |") 2. 358291 | ") in new stack"
-Execution [s@macro-dialout-trunk:1] Set ("SIP/205-087af3d8", "DIAL_TRUNK = 2") in new stack
-Execution [s@macro-dialout-trunk:2] GosubIf ("SIP/205-087af3d8","0?") Sub-pincheck | s | 1 ") in new stack"
-Execution [s@macro-dialout-trunk:3] GotoIf ("SIP/205-087af3d8","0?") disabletrunk | 1 ") in new stack"
-Execution [s@macro-dialout-trunk:4] Set ("SIP/205-087af3d8", "DIAL_NUMBER = 358291") in new stack
-Execution [s@macro-dialout-trunk:5] Set ("SIP/205-087af3d8", "DIAL_TRUNK_OPTIONS = tr") in new stack
-Execution [s@macro-dialout-trunk:6] Set ("SIP/205-087af3d8", "OUTBOUND_GROUP = OUT_2") in new stack
-Execution [s@macro-dialout-trunk:7] GotoIf ("SIP/205-087af3d8","0?") nomax ") in new stack"
-Execution [s@macro-dialout-trunk:8] GotoIf ("SIP/205-087af3d8","0?") chanfull ") in new stack"
-Execution [s@macro-dialout-trunk:9] GotoIf ("SIP/205-087af3d8","0?") skipoutcid ") in new stack"
-Execution [s@macro-dialout-trunk:10] Set ("SIP/205-087af3d8", "DIAL_TRUNK_OPTIONS =") in new stack
-[S@macro-dialout-trunk:11] Macro execution ("SIP/205-087af3d8","outbound callerid |") 2 ") in new stack"
-[S@macro-outbound-callerid:1] ExecIf execution ("SIP/205-087af3d8","0 |") SetCallerPres | ") in new stack
-[S@macro-outbound-callerid:2] ExecIf execution ("SIP/205-087af3d8","0 |") The value | REALCALLERIDNUM = 205 ") in new stack
-Execution [s@macro-outbound-callerid:3] GotoIf ("SIP/205-087af3d8","1?") normcid ") in new stack"
-Goto (macro-sortant-callerid, s, 6)
-Execution [s@macro-outbound-callerid:6] Set ("SIP/205-087af3d8", "USEROUTCID =") in new stack
-Execution [s@macro-outbound-callerid:7] Set ("SIP/205-087af3d8", "EMERGENCYCID =") in new stack
-Execution [s@macro-outbound-callerid:8] Set ("SIP/205-087af3d8", "TRUNKOUTCID = 383597") in new stack
-Execution [s@macro-outbound-callerid:9] GotoIf ("SIP/205-087af3d8","1?") trunkcid ") in new stack"
-Goto (macro-sortant-callerid, s, 12)
-[S@macro-outbound-callerid:12] ExecIf execution ("SIP/205-087af3d8","1" | ") The value | CALLERID (All) = 383597 ") in new stack
-Execution [s@macro-outbound-callerid:13] GotoIf ("SIP/205-087af3d8","1?") output ") in new stack"
-Goto (macro-sortant-callerid, s, 11)
-Execution [s@macro-outbound-callerid:11] MacroExit ("SIP/205-087af3d8","" "") in new stack
-[S@macro-dialout-trunk:12] ExecIf execution ("SIP/205-087af3d8","1" | ") AGI | fixlocalprefix") in new stack
-Launch Script AGI/var/lib/asterisk/acted-bin/fixlocalprefix
> fixlocalprefix: using chart 0 + [0] [9] X.
> fixlocalprefix: using mires NXXXXX 0 +.
== fixlocalprefix: Dialpattern 0 + NXXXXX matched. 358291-> 0358291
-Fixlocalprefix AGI completed Script, return 0
-Execution [s@macro-dialout-trunk:13] Set ("SIP/205-087af3d8", "OUTNUM = 0358291") in new stack
-Execution [s@macro-dialout-trunk:14] Set ("SIP/205-087af3d8", "custom = SIP/pstn") in new stack
-[S@macro-dialout-trunk:15] ExecIf execution ("SIP/205-087af3d8","0 |") The value | DIAL_TRUNK_OPTIONS = M(setmusic^) ") in new stack"
-[S@macro-dialout-trunk:16] Macro execution ("SIP/205-087af3d8","dialout-trunk-predial-crochet |" "") in the new battery
-Execution [s@macro-dialout-trunk-predial-hook:1] MacroExit ("SIP/205-087af3d8","" "") in new stack
-Execution [s@macro-dialout-trunk:17] GotoIf ("SIP/205-087af3d8","0?") bypass | 1 ") in new stack"
-Execution [s@macro-dialout-trunk:18] GotoIf ("SIP/205-087af3d8","0?") customtrunk ") in new stack"
-Execution [s@macro-dialout-trunk:19] Dial ("SIP/205-087af3d8","SIP/pstn/0358291 |") 300. ") in new stack"
-Called pstn/0358291
-SIP/pstn - 0885 has 138 sounds
-SIP/pstn - 0885 has 138 replied SIP/205-087af3d8
-Packet2Packet bypass 087af3d8/205-SIP and SIP/pstn - 0885 has 138
-Remote UNIX connection
-Connection to UNIX distance disconnected
-[H@macro-dialout-trunk:1] Macro execution ("SIP/205-087af3d8","hangupcall:" "") in the new battery
-Execution [s@macro-hangupcall:1] ResetCDR ("SIP/205-087af3d8", "w") in new stack
-Execution [s@macro-hangupcall:2] NoCDR ("SIP/205-087af3d8","" "") in new stack
-Execution [s@macro-hangupcall:3] GotoIf ("SIP/205-087af3d8","1?") skiprg ") in new stack"
-Goto (macro-hangupcall, s, 6)
-Execution [s@macro-hangupcall:6] GotoIf ("SIP/205-087af3d8","1?") skipblkvm ") in new stack"
-Goto (macro-hangupcall, s, 9)
-Execution [s@macro-hangupcall:9] GotoIf ("SIP/205-087af3d8","1?") theEnd ") in new stack"
-Goto (macro-hangupcall, s, 11)
-Execution [s@macro-hangupcall:11] Hangup ("SIP/205-087af3d8","" "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on "SIP/205-087af3d8" in the macro 'hangupcall '.
== Spawn extension h (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/205-087af3d8.
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/205-087af3d8"macro"dialout-trunk.
== Spawn extension (of-internal, 358291, 4) exited non-zero on 'SIP/205-087af3d8.

As you can see he composed, if found the dialing rule, but nothing happens on the SPA, it will not simply compose anything on.

I guess that something is wrong in RTC spa or configuration of trunk, but I don't know what. I tried like 5-6 guides of strach and none worked for me. Any suggestions?

Kind regards

Damir

HUH! I won't curse now, but I shoul!

Given that I'm NOT using any VOIP provider and I JUST wanted to use RTC it makes SENSE I don't connect anything to the WAN port, BUT when I configured the WAN port with my LAN IP (172.16.1.200) and put a few other IP on the LAN port, everything started working.

It's rly rly rly is NOT logical, but I got it working now.

Changed router and now I can't print

BT have changed my router. I use a Mac 10.9. My printer is an officeject 4620. It worked fine before the change in router. I can print to the printer itself, I can print using the HP software (but it ask me to enter the IP address and then enter the router password) but cannot print from any software, that is to say the word. I don't get an error message and the computer seems to try and send it to the printer, but it does not. Someone at - it ideas?

Hi lynneseymour,

Welcome to the HP Forums!

I read your post about not being able to print wireless from the evolution of your router, and I'd be happy to help!

With regard to the new router you use, what's his name, and model number? How far is the router of the printer? There is a parameter called "Multicast" router, and it must be enabled for printing from a Mac.

To connect the printer to the new network, you will need to restore the default network settings on the front panel.

This can be done by following these steps:

(1.) by clicking on the key icon on the lower part of the front panel

2.) click the arrow on the button down to network and press Ok

3.) scroll down to wireless settings and then click Ok

Scroll 4) to the network by default and press Ok

5.) click Yes and the default network will be restored

Then, the printing system needs to empty the old print queue. To do this, follow this document about the reset printing system.

Do not click on the button to add another printer, the next step is to run the HP installation wizard to perform an automatic wireless connection.

The next step is to run the HP Setup Assistant to start an automatic wireless connection with a usb cable.

Follow the steps under "optional: I installed my printer... to move to a wireless network connection" to install the software of the printer for a wireless network connection.

I will be looking forward to meet you and thank you for visiting the HP Forums!

Problem with GCE international calls

Hi all

I have my router CME with two different link:

the first is the VOIP and the second is POTS

I need to allow international outgoing calls be routed via the link of POTS

So I made these commands:

voice pots Dial-peer 5
corlist outgoing international call
International call through TT description
preference 1
destination-model 000 t
port 0/0/0:15

I also configured the international calls through SIP TRunk by using these commands

Dial-peer voice voip 2200
corlist outgoing international call
Description * International Outgoing Call to SIP Trunk *.
translation-profile outgoing PSTN_Outgoing
destination-model 000 t
session protocol sipv2
session target ipv4:192.168.2.51
codec voice-class 1
voice-class sip dtmf-relay rtp - nte force
DTMF-relay rtp - nte
No vad

The problem is that calls always goes in the SIP trunk while I've specified the upper value of the preference for POTS

Y at - it all missconfiguration

Please help me it is urgent!

Add "preference 2" under the VoIP dial peer. Please note that when you set the order of preference, the lower part, number preferably, more priority. Absolute priority is given to the counterpart of dial with preference order 0 and it is the default value.

Manish

-Does the rate of useful messages

Typical call for CM flow / unit / Fax without obtaining additional?

Configuration:

Centralize the VMO unit / Call Manager / fax server in the data center. Routers SRST supporting all offices. Of Fax Server supported for the unit, such as RightFax or others on the approved list.

What would the typical call flow if we wish to have the DID numbers subscribed to do double-duty as their fax number (so that we don't need to get all the new lines DID new business cards, etc.)

Is it still possible? How would an incoming fax call get routed to the fax server automatically rather than as an incomprehensible series of fax tones in the voice mail box?

What happens if the user is in fact on their phone and hear the fax tone? Is it possible to Call Manager program to generate a soft button on the 7960 that says 'Send fax'?

Thank you.

Short story is there is no way to achieve smooth - for reasons that you kind of point in your question (i.e., what to do if the user is on their phone?). Correctly do gateways would have generated fax tone detection (in some cases, the sender Server/fax won't that until he hears a fax tone on the side response - even if it has a hole in it) and spend an event upstairs that is supplied to the unit via CM - us would then is a fax call and send it to the fax server as well as information on the phone number and number composed as the fax ended up being routed to the right area.

There was people who tried to do things moderately awkward IVR on gateways (i.e. the caller hears "if it is a voice call, press 1" which has then passes the call via as usual or he will send the call to a port on the fax)-not an ideal interface at all and the location of is not possible among other problems.

Is really the only way to go about it.

VPN tunnel between the concentrator 3005 and router Cisco 827

I am trying to establish a VPN tunnel between the Central Office with VPN 3005 and controller branch Cisco 827 router.

There is a router of perimeter with access set up in front of the 3005 list.

I quote the ACLs on the Central perimeter router instructionsuivante to allow traffic to permit ip 3005 - acl 101 all 193.188.X.X (address of the hub)

I get the following message appears when I try to ping a local host in the Central site.

Can Anyoune give me the correct steps to 827 and 3005.

Thank you

CCNP Ansar.

------------------------------------------------------------------------------------------------------

Debug crypto ISAKMP

encryption of debugging engine

Debug crypto his

debug output

------------------

1d20h: IPSEC (sa_request):,.

(Eng. msg key.) Local OUTGOING = 172.22.113.41, distance = 193.188.108.165.

local_proxy = 202.71.244.160/255.255.255.240/0/0 (type = 4),

remote_proxy = 128.128.1.78/255.255.255.255/0/0 (type = 1),

Protocol = ESP, transform = esp - esp-md5-hmac.

lifedur = 3600 s and KB 4608000,

SPI = 0x83B8AC1B (2209917979), id_conn = 0, keysize = 0, flags = 0x400D

1d20h: ISAKMP: ke received message (1/1)

1d20h: ISAKMP: 500 local port, remote port 500

1d20h: ISAKMP (0:1): entry = IKE_MESG_FROM_IPSEC, IKE_SA_REQ_MM

Former State = new State IKE_READY = IKE_I_MM1

1d20h: ISAKMP (0:1): early changes of Main Mode

1d20h: ISAKMP (0:1): lot of 193.188.108.165 sending (I) MM_NO_STATE

1d20h: ISAKMP (0:1): retransmission phase 1 MM_NO_STATE...

1d20h: ISAKMP (0:1): will increment the error counter on his: retransmit the phase 1

1d20h: ISAKMP (0:1): retransmission phase 1 MM_NO_STATE

1d20h: ISAKMP (0:1): lot of 193.188.108.165 sending (I) MM_NO_STATE

1d20h: ISAKMP (0:1): retransmission phase 1 MM_NO_STATE...

1d20h: ISAKMP (0:1): will increment the error counter on his: retransmit the phase 1

1d20h: ISAKMP (0:1): retransmission phase 1 MM_NO_STATE

1d20h: ISAKMP (0:1): lot of 193.188.108.165 sending (I) MM_NO_STATE

1d20h: IPSEC (key_engine): request timer shot: count = 1,.

You must also allow the esp Protocol in your ACL.

access-list 101 permit esp any host x.x.x.x (address of the hub)

Hope this helps,

-Nairi

ALTER audio during phone call

Hello

I would like to know if this is possible and how it is possible. I need to put in place a mechanism by which a user can activate an application and make a phone call and then the audio on both sides of the call is routed to the application, the application modifies the audio and sends it back. This may seem crazy or stupid, but we have a legitimate and serious company for this requirement. I also need a way to do the same thing for an incoming call.

I would like to know if there is an available API (is it is limited or licensed API an arrangement is possible) or no work around if there is not.

Concerning

Lionel

It is no longer possible.

The other limitation that I see would be the choice of phone.

I know that CDMA phones cannot record audio during audio playback, so these phones would have to do more a walkie talkie interface - you can talk or listen, but not both at the same time.

I'm not sure if the GSM phones have this limitation as well

Branch call PSTN routing

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