Transfer_redirect: Failed to parse destination URI

Asterisk Asterisk SIP
1

Hi,

trying a simple blind transfer on incoming call, but i get error below? what am i doing wrong?
6000 is calling to 123 and i want to blind transfer to an linhome extension

[Feb 19 10:53:13] WARNING[399]: chan_pjsip.c:1916 transfer_redirect: Failed to parse destination URI 'outgoing-flexisip/sip:xxx@sip.linhome.org' for channel PJSIP/6000-00000001
[Feb 19 10:53:13]     -- Auto fallthrough, channel 'PJSIP/6000-00000001' status is 'UNKNOWN'

exten => 123,1,Transfer(PJSIP/outgoing/sip:xxx@sip.linhome.org)

and in pjsip.conf:

[outgoing]
type=endpoint
disallow=all
allow=ulaw,alaw
allow=h264
from_domain=asterisk.com
2

Transfer always uses the incoming session, so doesn’t need to know the end point and the technology, if supplied, is only used to prevent transfers on inappropriate ones.

3

So I don’t need to specify the outgoing ?

4

if i do without outgoing like below, i receive:

[Feb 19 14:05:55] NOTICE[397]: res_pjsip_session.c:3980 new_invite: 6000: Call (UDP:192.168.0.232:5060) to extension 'xxx' rejected because extension not found in context 'default'.

exten => 321,1,Transfer(PJSIP/sip:xxx@sip.linhome.org)

not sure why it doesnt find xxx @ that linhome service?

if i do this, with a dial command it works:

Dial(PJSIP/outgoing/sip:xxx@sip.linhome.org)

5

What is 192.168.0.232? What is actually in the Refer-To header? It appears that the imediately upstream system has sent the call back to you.

6

Hi, that’s endpoint 6000, registered on sofrphone callling to 123

7

The softphone probably doesn’t know how to send calls other than through Asterisk, so ignores the domain name. (More specifically, Asterisk may be the phones outbound proxy and Asterisk normally ignores domain names and assumes all requests are aimed at it.

8

So how can I fix/force this? :frowning:

9

Why are you even trying to do it? You would need to provide all the credentials to the phone in order for it to successfully initiate a direct call to the provider. In any case, that is a question about the phone.

If you are going to go through Asterisk anyway, there is no point in doing the transfer, so even to the extent that it might be possible to have Asterisk distinguish between local and remote domains, there would appear to be no point in doing anything other than connecting directly to the new destination.

10

Hmm, for Linhome service, no credentials are required, it’s unauthenticated… If I do the same with the dial application, with 6000 then it works…

I want to have asterisk listening… I can receive an invite from my door intercom and want to transfer it to a Linhome user externally

11

Dial initiates a call from Asterisk to Linhome. Transfer attempts to initiate a call from the softphone to Linhome, but it looks like the softphone is configured (or hard coded) to send all calls via an outbound proxy, which is Asterisk, so it sends the call back to Asterisk, but Asterisk has no extension defined for calls to xxx, or at least not in the context in which the softphone is making the call.

In any case, the only way that Asterisk could handle that call would be to issue a DIal() command, so, again, one has to ask the question why are you going through the phone, when you could directly issue the Dial command.

12

Yeah, I’m still missing video if I do the dial command, that’s why I wanted to tryout the transfer…

Gonna do a test again without the softphone…
Thnx in advance

13

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.