Call Transfer to external number

Hi All,

I have an issue related to call transfer to Tech Support in IVR. While the call comes to IVR i.e. on DID number from external number, the IVR prompts are playing. In between the IVR we have the option to transfer to Tech Support, means the call should transfer to Tech Support. But the call is not transferring to Tech Support. We are using the following code to transfer the call

exten => _X.,1,Set(CALLERID(num)=9852XXXXXX)
exten => _X.,n,Dial(SIP/9852XXXXXX/9853XXXXXX)

When we are registering the Soft-phone to the asterisk server with same DID(9852XXXXXX) and calling to IVR, at that time we are able to transfer the call to Tech Support. But when we are registering the Soft phone with any number other than DID or we are calling from any other external number to IVR, at that time we are not able to transfer the call to Tech Support.

Here are the logs when we are registering the soft-phone with DID number and calling to IVR.

– Executing [_X.@TechSupport:2] Dial(“SIP/9852XXXXXX-000001e0”, “SIP/9852XXXXXX/9853XXXXXX”) in new stack
== Using SIP RTP CoS mark 5
– Called 9852XXXXXX/9853XXXXXX
[Jul 3 05:04:50] WARNING[18640]: chan_sip.c:12799 __set_address_from_contact: Invalid host name in Contact: (can’t resolve in DNS) : ‘9852XXXXXX’
– SIP/9852XXXXXX-000001e1 answered SIP/9852XXXXXX-000001e0
– Packet2Packet bridging SIP/9852XXXXXX-000001e0 and SIP/9852XXXXXX-000001e1
== Spawn extension (TechSupport, _X., 2) exited non-zero on 'SIP/9852XXXXXX-000001e0’
ivr-01*CLI>

The logs when we are calling from Soft-phone which is registered with 2000 to asterisk server.

– Executing [_X.@TechSupport:2] Dial(“SIP/2000-000001e3”, “SIP/9852XXXXXX/9853XXXXXX”) in new stack
== Using SIP RTP CoS mark 5
[Jul 3 06:03:49] WARNING[30495]: app_dial.c:1780 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/2000-000001e3’ status is 'CHANUNAVAIL’
ivr-01*CLI>

When the call will come to IVR from any external number then in logs the external number will appear in stead of 2000.

I need your help to resolve this issue. Please help.

Thanks

[quote=“pankajthakur”][Jul 3 05:04:50] WARNING[18640]: chan_sip.c:12799 __set_address_from_contact: Invalid host name in Contact: (can’t resolve in DNS) : ‘9852XXXXXX’
[/quote]

You don’t have a section labelled 9852XXXXXX in your sip.conf!

More generally, this is the wrong forum for support questions and, as Asterisk itself doesn’t have an IVR concept, you need to talk to the person who implemented that in your dialplan, or use a forum for the add on software that imlemented the concept.

On second thoughts, the problem is that the incoming Contact header is invalid (no domain name).

I don’t understand why the two cases differ, but I do wonder whether you are using Asterisk “realtime”. The error message in the second one indicates that either there is no sip.conf entry; or its registration has expired (unlikely that you would register a DID incoming); or you have qualify set and the qualify has timed out.

Hi david55
Thanks for your reply.

Here is the Sip configuations for the DID

[general]
context=default
bindaddr=XXX.XXX.XX.XXX
port=5060
allowguest=yes ; Allow or reject guest calls (default is yes)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=alaw ; Allow codecs in order of preference
language=en ; Default language setting for all users/peers

register => 9852XXXXXX:XXXXXXXX@sbc-met.telco.bz/9852XXXXXX
;============================================
[9852XXXXXX]
username=9852XXXXXX
type=friend
secret=XXXXXXXX
outboundproxy=sbc-met.telco.bz
nat=yes
;directmedia=yes
;insecure=very
insecure=invite
;allowguest=yes
host=as1.telco.bz
;host=dynamic
;fromdomain=9852XXXXXX
fromdomain=as1.telco.bz
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
;allow=g729
context=default
;context=custom-get-did-from-sip
authname=9852XXXXXX
canreinvite=yes
;srvlookup=yes
qualify=yes

[1000]
type=friend
username=1000
callerid=1000
host=dynamic
secret=XXXX
canreinvite=yes
dtmfmode=rfc2833
;canreinvite=yes
nat=yes
outboundproxy=sbc-met.telco.bz
;qualify=yes
context=default
disallow=all ; First disallow all codecs
;allow=g729 ; see doc/rtp-packetization for framing options
allow=ulaw ; Allow codecs in order of preference
allow=alaw ; Allow codecs in order of preference
qualify=yes

[2000]
type=friend
username=2000
host=dynamic
secret=XXXXX
canreinvite=yes
dtmfmode=rfc2833
nat=yes
qualify=yes
context=default
disallow=all ; First disallow all codecs
;allow=g729 ; see doc/rtp-packetization for framing options
;callerid=2000
allow=ulaw ; Allow codecs in order of preference
allow=alaw ; Allow codecs in order of preference
authname=2000

Here is some more cli track.

ivr-01CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sbc-met.telco.bz:5060 N 9852XXXXXX 105 Registered Fri, 06 Jul 2012 02:48:40
1 SIP registrations.
ivr-01
CLI>
ivr-01CLI>
ivr-01
CLI>
ivr-01CLI>
ivr-01
CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
1000/1000 (Unspecified) D N 5060 UNKNOWN
2000/2000 XXX.XXX.XX.X D N 28253 Unmonitored
9852XXXXXX/9852XXXXXX XXX.XX.XXX.X N 5060 UNREACHABLE
3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline]

Now I am not sure whether the problem is with SIP configuration or some where else.
Please reply. Its a bit urgent.

Thanks

Turn off qualify. You have an unreachable status which means that qualify has failed the connection. Eitehr their system is dropping the OPTIONS requests used to test the link, or they are getting dropped somewhere else.

Why do you have allowguest (security risk, and can confuse problems with misconfiguerd incoming trunks)?

Why do you have nat=yes (it does not do what people seem to think it does)?

Why do you have type=friend (type=peer is normally sufficient and is a lot more secure)?

Hi david55

The issue is resolved. The only thing need to add was Set(CALLERPRES()=allowed) in extensions.conf before setting the callerid and in SIP.conf canreinvite=no

Thanks for your reply.

Hi All,

I am able to transfer the call successfully from DID to external number ie. from 9852XXXXXX to 9853XXXXXX using (SIP/9852XXXXXX/9853XXXXXX). At this number another IVR is playing and in that IVR when I am entering the extension 211 then I am able to connect to customer care.

But I want to transfer the call directly to customer care ie. to extension 211. I am not getting how to do this.
Please help.

anyone tell me the way how can i transfer our call to other number without showing our number.

Hi,

can anyone tell me how to transfer a call directly to extension on remote server.
I have two DID(9852XXXXXX) and 9853XXXXXX on two remote servers. I want to transfer the call from 9852XXXXXX to extension 211 which is on second remote server having DID(9853XXXXXX). I am able to transfer call from 9852XXXXXX to 9853XXXXXX by using Dial(SIP/9852XXXXXX/9853XXXXXX). but how to transfer directly to extension 211.

Please somebody suggest how to do this.

Thanks in advance.

Try:

Hi dejanst.

Thanks for the reply. This problem is solved.

Can we check in asterisk, the number of channels in SIP trunk provided by trunk provider.

Glad to hear that I helped to solve your problem :wink:

The maximum number of concurrent channels for a SIP trunk can not be seen on Asterisk. Your provider knows that information :wink:

The provider might not set an explicit limit. It might also be set by the capacity of your internet connection and by the conflicting traffic, over that, at the time. It might be set at the ITSP, but as an aggregate over all their users, based on the number of ISDN channels that they have.

In all cases, nothing provides that information to Asterisk.

Hi All

I need your help again. I am making the outbound call for one number (9853XXXXXX) and I am transferring the call to another number(9852XXXXXX). Now what i want is, I want to display the information on the softphone ie. 9852XXXXXX. The information is like CallerName, CallerID, Location.
I am able to display CallerName and CallerID. But I dont know how to transfer the other data during call transfer or whether is it possible or not.
Is there any function or method by which we can transfer the data at call transfer and can display on the softphone(9852XXXXXX).

Need your help.

Thanks