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

Nicolas Dramais ndramais at indigosw.com
Wed Dec 3 11:04:51 MST 2003

Hi Pertti and all, Thank you for your reply Pertti! Latest CVS doesn't 
work for us and we've tried to downgrade back to CVS-09-29 as you 
mentioned but with no success with our setup. I would be highly 
interested in knowing what other experts on the list think about this 
issue. Mark, I know you're highly sollicited, but could you please have 
a look at this issue and tell us what you think should be done ? Thank 
you very much in advance and long life to Asterisk With best regards, 

Pertti Pikkarainen wrote:

    I have seen this just recently with a setup that is quite close to
    the one you are using. The problem happened when I tried to upgrade
    all: Zaptel, Libpri and Asterisk. PSTN -----Asterisk
    ---[existingPBX] | SIP PSTN <-> SIP Worked ok both ways. PSTN <- *
    <- PBX The same problem ( Missing mandatory IE ... ) At that time I
    downgraded back to CVS-09-29 and got it working again. Probably
    Libpri would have been enough. Then again I have exactly the same
    setup elsewhere where the PBX is running the latest software. There
    I'm able to use the latest CVS. --Pertti 

Nicolas Dramais wrote:

>> 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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