[Asterisk-Dev] E1/PRA connection to existing PBX with Asterisk - SETUP ACK with Suggested Channel ID missing ?

[Nicolas Dramais]

Dear Asterisk experts,

I would like to draw your valuable attention on the following situation 
to see whether some of you have encountered it before and have found 
solutions / workarounds.

We want to place Asterisk behind an existing PBX (connected to the PSTN) 
using an E1/PRA line.

PSTN -- Telco E1 -- [existing PBX] -- E1/PRA -- [Asterisk+Digium E100P] 
-- SIP phones


We observe the following behaviour:

    * in the call direction "Asterisk" to "existing PBX" (SIP phone
      calling out to PSTN), it works great
    * in the call direction "existing PBX" to "Asterisk" (PSTN user
      calling in to SIP phone), we get the following error on the
      Asterisk console:

WARNING[11276]: File chan_zap.c, Line 5719 (zt_pri_error): PRI: XXX
> >>>>Missing mandatory IE 24/Channel Identification XXX

    * in the call direction "existing PBX" to "Asterisk", we observe
      that the Q.931 SETUP message sent by the "existing PBX" doesn't
      contain a channel identification IE

We double-checked the Q.931 SETUP format on DSS1 specification and we 
found that the Channel Identification is: "Mandatory in the 
network-to-user direction. Included in the user-to-network direction 
when a user wants to indicate a channel. If not included, its absence is 
interpreted as 'any channel acceptable'." (section 3.1.14, note 4).

We assume that the call direction "existing PBX" -> "Asterisk PBX" is 
"user-to-network", which means that the channel identification IE is 
optional in SETUP messages sent by the "existing PBX", meaning 'any 
channel acceptable'.

(Asterisk is configured to play the NETWORK side with 
"signalling=pri_net" in zapata.conf)


However, Asterisk doesn't seem to want to select a channel and return it 
to the "existing PBX" in the SETUP ACK (as does a Cisco gateway in the 
exact same configuration).

In conclusion, it seems that the functionality to answer a call without 
a given channel is not there in Asterisk and we think missing this 
functionality is critical as we believe the above scenario is an 
important and likely usage scenario for many people.

Can someone please confirm the above observation ?
If yes, does anyone (Mark ?) have plans to add that functionality into 
Asterisk in the near future ?

We'd be happy to help beta test it .

Thank you all in advance for your answers and thank you all anyhow for 
the great work you've done with Asterisk.

Best regards,

Nicolas Dramais - Belgium
 


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