[Asterisk-Users] One-way talk paths (without INVITE?) and other issues

[Moshe Yudkowsky]

I'm experiencing one-way voice paths, followed by a hangup on one 
softphoine and not the other. Also, caller ID has odd outputs -- and I 
wonder if the problems are related.

My configuration has Asterisk and a Linphone softphone running on the 
same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect 
to the Linphone instance.

When I call from the PC to Linphone:

* I call from the PC (user m12) to Linphone (usr m4), which rings

* I answer on Linphone

* Asterisk attempts to set up a talk path. Here's the output from 
Asterisk, with Linphone (m4) connecting to the PC (m12):

>     -- Called m4
>     -- SIP/m4-8f2b is ringing
>     -- Registered SIP 'm12' at 172.28.54.34 port 5060 expires 500
>     -- SIP/m4-8f2b answered SIP/m12-195f
>     -- Attempting native bridge of SIP/m12-195f and SIP/m4-8f2b


I don't know how the "registered" statement appreared in the middle.

At this point I can talk into the PC and hear it on Linphone -- but I 
cannot speak into Linphone and hear myself on the PC.  After about 10 
seconds, possibly less:

* The PC phone gets a hangup message (BYE).

* Linphone does *not* get a hangup message and remains offhook. Any 
attempt to call Linphone from the PC results in Asterisk routing the 
call to voicemail. I must manually hang up Linphone.

Oddly enough, the caller ID displayed by Linphone, and apparently sent 
by Asterisk, is incorrect. Instead of showing "m12" as the caller ID, 
Linphone receives "m1" as the caller ID:


> INVITE sip:m4 at x1.x2.x3.x4:5062 SIP/2.0
> Via: SIP/2.0/UDP x1.x2.x3.x4:5060;branch=z9hG4bK0c4d7e4c
> From: "m1" <sip:asterisk at x1.x2.x3.x4>;tag=as62b91e33
> To: <sip:m4 at x1.x2.x3.x4:5062>;tag=4210403538
> Contact: <sip:asterisk at x1.x2.x3.x4>
> Call-ID: 478c64565bea701e0f3eb830731011a5 at x1.x2.x3.x4
> CSeq: 104 INVITE
> User-Agent: Asterisk PBX
> Content-Type: application/sdp
> Content-Length: 159

(x1.x2.x3.x4 substituted for the actual IP address.)

To my fairly untrained eye, this looks like a legitimate proxy message 
but the caller ID is wrong. My SIP configuration file does contain both 
m1 and m12 as legitimate callers:

 > [m1]
 > type=friend
 > username=m1
 > host=dynamic
 > permit=x1.x2.x3.0/16

 > [m12]
 > type=friend
 > username=m1
 > host=dynamic
 > permit=x1.x2.x3.0/16

I also note the following Asterisk warnings. I cannot tell if they 
happen just before or just after I lose the one-way voice path:

> WARNING[81926]: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call 478c64565bea701e0f3eb830731011a5 at 172.28.54.160 for seqno 103 (Request)
> WARNING[81926]: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call 478c64565bea701e0f3eb830731011a5 at 172.28.54.160 for seqno 104 (Request)

Furthermore, I have yet to see a SIP channel disappear after a call is 
over. They are always listed as active, even hours later. Here are is 
the result of "show sip channels":

> Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  Format
> 172.28.54.160    m4          478c64565be  00104/00000  00000ms  0000ms  0
> 172.28.54.160    m4          4d330ced01e  00104/00001  00000ms  0000ms  0
> 172.28.54.160    m4          0bb7855f15b  00104/00001  00000ms  0000ms  0
> 172.28.54.160    m4          3db8538b4b4  00104/00001  00000ms  0000ms  0
> 4 active SIP channel(s)


but none of these calls are "active" in any sense of the term that I can 
think of. I have tried to use the "sip show channel" command for further 
testing, but apparently I don't understand the syntax of the command -- 
"sip show channel 478c64565be" and "sip show channel m4/478c64565be" and 
"sip show channel SIP/m4-8f2b" all give the same error message, "no such 
SIP call ID 'whatever'". Either I don't understand the "sip show 
channels" command, or there's a bug.

My questions are:

* How is it that I get one-way voice paths? Is this a configuration 
problem? Are the INVITES not getting through but the voice paths 
established anyway?

* What's the problem with the incorrect caller IDs? I have *no* caller 
ID settings that I'm aware of in my *.conf files. The PC's program 
registers as "m12" but Asterisk sends "m1" as the name. PC-side 
debugging shows that the PC sends "m12 at whatever" as its name.

Although this feels like a bug, I strongly suspect that I'm missing some 
simple SIP configuration issue, but I haven't been able to track it down 
just yet.  And I'd like to clarify any other issues before starting on a 
bug hunt.

-- 
  Moshe Yudkowsky * http://www.Disaggregate.com