[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,
Nicolas
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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