* You are viewing Posts Tagged ‘outbound route’

OUTBOUND and INBOUND routes

Hello All,

I have a pstn line can have the local, STD and ISD
capabilities. My local number is 91471-2527XXX and the region is India. I
would like to use the number for all possible calls ( local, STD and ISD
call facilities to Land line and mobile phones) through an FXO card
configured in asterisk freepbx.

Can anybody help me to create an outbound route and inbound route required
in freepbx for the above requirement ?

Thanks,
Michael.k

I can’t figure out how to redirect a call to a trunk.

OK, i am hoping that someone will be able to help me out.
I am using FreePBX 2.8.1.4

I have two asterisk servers connected with a iax trunk.
The trunk is working fine when used via the outbound route setting.
meaning an extension on one server can call a specific extension on
the other server.

now what i want to do is set it up so that an incoming call (from a
third server)
is redirected to the second server from the first server when the
extension number does exist on the first server.
I thought I knew how to do this…. setup an inbound route that
answers any inbouond call and set the desitnation
to be “trunk” “server2″

but this doesn’t seem to be working.
I know the inbound route is being called, i can even see the first
server trying to forward the call to server 2.
but the problem seems to be that no $OUTNUM is being set, so no
extension number on the second server is specified…
at which point the call fails. I get an “all circuits are busy now”
announcement.

i upgraded to freepbx 2.8.x.x because of the fact that “FOLLOWME” and
“INBOUND ROUTES” were both able to specify a trunk
as a desitnation… but neither are working for me now?

am i doing something wrong?

Avaya & Asterisk FreePBX Integration Problem

Hi,

I’m currently testing my FreePbx Box to work with our Avaya PBX to allow
dialing outgoing international call and FreePBX extensions to avaya PBX
Extensions calling.
Unfortunately no luck to do it successfully. Any help would be much be
appreciated, here is the sample codes I already tried:

On FreePBX GUI:
1. I created a custom Trunk called AvayaPBXTrunk with custom dial string
OOH323/$OUTNUM$/Avaya
2. Created an Outbound route called InternationalCall and select
AvayaPBXTrunk on the trunk sequence.
3. Created an Extension 1000 with dial extension OOH323/$OUTNUM$@Avaya

On Asterisk CLI:
1. Edit ooh323.conf with the following codes:
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=127.0.0.1
port=1720
callerID=”Asterisk PBX”
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
callerid=asterisk
context=default
disallow=all
allow=ulaw

[Avaya]
type=friend
context=from-internal
host=X.X.X.X ‘IP Address of our Avaya PBX
port=1720
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=inband
rtptimeout=60
e164=50

2. Edit sip_custom.conf with the following code:

[general]
context=from-internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=127.0.0.1
srvlookup=yes
canreinvite=no

Below also the log result during the call:

DTMF not being heard correctly by far end conference system

Am 12.01.2011 11:37, schrieb Duncan Turnbull:
> Hi there
>
> I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can’t nail down the reason. I don’t believe this is a general issue, rather some specific conference systems that they need.
>
> I am sure I saw this covered a few years ago but can’t find it in the lists.
>
> The phones and the system are using rfc2833 and either alaw or ulaw, I have stayed away from in band dtmf, but may need to consider it. They also use *1 to turn on call recording and I am not sure how that will go with inband.
>
> Another 1.6 system has no problem with being detected and it uses SIP trunks from the same supplier as the customer.
>
> The first system is a 1.4.38 box, it has sip trunks as the primary outbound route, the secondary route is iax to another box then via analogue lines. Almost all the handsets are sip and a re a mix of polycom and yealink.
>
> The sip trunks routed out through the iax link via analogue lines seem to work okay too. I am wondering if the iax handling of dtmf matches whatever the far end is expecting a little better
>
> For now I have routed everything via the iax / analogue lines which may cause some problems in terms of line availability but gets past the issue. I am considering upgrading the box to 1.6 as the working one is 1.6
>
> The other box is a digium AA50 appliance so I can’t do much with it, other than find the right settings.
>
> I have on the first one
> relaxdtmf=yes - relates to old issues too as far as I can tell
> rfc2833compensate=yes - this only appears to matter for inbound
>
> I’m not sure these do anything useful
>
> > From what I can tell it could be the toneduration, but don’t know what it should be, and while technically its probably the IVR being fussy that doesn’t help me and I want to see why one system works and one doesn’t
>
> This is dtmf debug from an iax handset sending digit 4
>
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-00000639 to write format slin
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 160 sample intervals
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:5297 ast_channel_start_silence_generator: Started silence generator on ‘SIP/xtreme-00000639′
> [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2796 ast_rtp_raw_write: Difference is 1736, ms is 237
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 0 sample intervals
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:5310 ast_channel_stop_silence_generator: Stopped silence generator on ‘SIP/xtreme-00000639′
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-00000639 to write format alaw
> [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 1713844722 to 565606422 due to a source change
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF begin on channel (IAX2/419-13088)
> [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-00000639
> [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 565606422 to 226872656 due to a source change
> [Jan 12 23:13:56] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF end on channel (IAX2/419-13088)
> [Jan 12 23:13:56] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update
> [Jan 12 23:13:56] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-00000639
> [Jan 12 23:13:56] DEBUG[8717]: res_features.c:1399 feature_interpret: Feature interpret: chan=IAX2/419-13088, peer=SIP/xtreme-00000639, code=4, sense=1
>
> I will get a sip dump but am remote for now and don’t have sip access
>
> All pointers and knowledge appreciated
>
> Cheers Duncan
As far as I can remember you should take a look at the used codec and
this here:
http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode

Some codecs do not feel happy with some seetings for dtmfmode. Perhaps
you may comapre these on your 2 boxes.

DTMF not being heard correctly by far end conference system

Hi there

I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can’t nail down the reason. I don’t believe this is a general issue, rather some specific conference systems that they need.

I am sure I saw this covered a few years ago but can’t find it in the lists.

The phones and the system are using rfc2833 and either alaw or ulaw, I have stayed away from in band dtmf, but may need to consider it. They also use *1 to turn on call recording and I am not sure how that will go with inband.

Another 1.6 system has no problem with being detected and it uses SIP trunks from the same supplier as the customer.

The first system is a 1.4.38 box, it has sip trunks as the primary outbound route, the secondary route is iax to another box then via analogue lines. Almost all the handsets are sip and a re a mix of polycom and yealink.

The sip trunks routed out through the iax link via analogue lines seem to work okay too. I am wondering if the iax handling of dtmf matches whatever the far end is expecting a little better

For now I have routed everything via the iax / analogue lines which may cause some problems in terms of line availability but gets past the issue. I am considering upgrading the box to 1.6 as the working one is 1.6

The other box is a digium AA50 appliance so I can’t do much with it, other than find the right settings.

I have on the first one
relaxdtmf=yes - relates to old issues too as far as I can tell
rfc2833compensate=yes - this only appears to matter for inbound

I’m not sure these do anything useful

From what I can tell it could be the toneduration, but don’t know what it should be, and while technically its probably the IVR being fussy that doesn’t help me and I want to see why one system works and one doesn’t

This is dtmf debug from an iax handset sending digit 4

[Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-00000639 to write format slin
[Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 160 sample intervals
[Jan 12 23:13:55] DEBUG[8717]: channel.c:5297 ast_channel_start_silence_generator: Started silence generator on ‘SIP/xtreme-00000639′
[Jan 12 23:13:55] DEBUG[8717]: rtp.c:2796 ast_rtp_raw_write: Difference is 1736, ms is 237
[Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 0 sample intervals
[Jan 12 23:13:55] DEBUG[8717]: channel.c:5310 ast_channel_stop_silence_generator: Stopped silence generator on ‘SIP/xtreme-00000639′
[Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-00000639 to write format alaw
[Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 1713844722 to 565606422 due to a source change
[Jan 12 23:13:55] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF begin on channel (IAX2/419-13088)
[Jan 12 23:13:55] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update
[Jan 12 23:13:55] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-00000639
[Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 565606422 to 226872656 due to a source change
[Jan 12 23:13:56] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF end on channel (IAX2/419-13088)
[Jan 12 23:13:56] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update
[Jan 12 23:13:56] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-00000639
[Jan 12 23:13:56] DEBUG[8717]: res_features.c:1399 feature_interpret: Feature interpret: chan=IAX2/419-13088, peer=SIP/xtreme-00000639, code=4, sense=1

I will get a sip dump but am remote for now and don’t have sip access

All pointers and knowledge appreciated

Cheers Duncan