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Summary:ASTERISK-06596: Polycom+Snom cannot answer calls when sip.conf [general] callerid="Seemingly Legit" <5551212>
Reporter:khmann (khmann)Labels:
Date Opened:2006-03-22 11:21:17.000-0600Date Closed:2006-05-31 09:00:20
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip-callerid-failed.txt
Description:When specifying a callerid (sip.conf [general]) to override the default "asterisk" for anonymous calls,

a Polycom IP600 (multiple firmware versions) is unable to answer the call.
a Snom 190 (latest firmware) is unable to transfer the call after answering.

****** ADDITIONAL INFORMATION ******

remove the callerid= entry, and everything is fine.
This problem was occuring under v1.0 and also occurs under 1.2.5.
Comments:By: Olle Johansson (oej) 2006-03-22 11:26:44.000-0600

Ok, the bug guidelines clearly states that you need to upload a SIP DEBUG of failed SIP transactions. Please follow them, otherwise it will be very hard for me to make any useful comment at all.

Thanks!

/O
By: Joshua C. Colp (jcolp) 2006-03-22 11:28:28.000-0600

Use the following instead:

callerid=Seemingly Legit <5551212>

The start and end " are used in the callerid, and can end up creating corrupted SIP invites which freak phones out.
By: Olle Johansson (oej) 2006-03-22 11:29:47.000-0600

I still want to see a corrupted SIP invite :-)
By: khmann (khmann) 2006-03-22 16:11:38.000-0600

Sorry for blatantly ignoring the bug guidelines, a SIP debug is now attached showing the unattainable call.

Interestingly, joshnet's suggestion of
callerid=Seemingly Legit <5551212>
didn't work any different, but
callerid=Seemingly Legit
does work.  hmmmm.
By: Olle Johansson (oej) 2006-03-23 00:54:12.000-0600

Hmmm. Something is wrong in the parsing since we include the <5551212> within the display name. Are you sure this was the exact syntax used in sip.conf?

callerid="Seemingly Legit" <5551212>
By: khmann (khmann) 2006-03-24 09:31:32.000-0600

100%.  just copied and pasted.  this behavior is also in 1.0.
By: opsys (opsys) 2006-04-30 01:12:34

Is this still a problem??

Can you run show channel {CHANNEL of Polycom or SNOM} and post.
By: khmann (khmann) 2006-05-01 08:56:30

Just tested with asterisk-1.0.7-1, issue still exists.  Callerid specified in [general] section of sip.conf as:

callerid="Seemingly Legit" <5551212>

Phones correctly display both the name and the number.  I have tried various forms of the above with the same results.

Polycom IP600 (SIP.ver=1.6.2.0041 + other older) is unable to answer the call.
Snom 190 (current 3.56z? + other 3.5 tested) is able to answer, but unable to transfer (call is put on hold locally, can be retrieved locally)

asterisk*CLI> show channel SIP/Chrissy-ffcd
-- General --
          Name: SIP/Chrissy-ffcd
          Type: SIP
      UniqueID: 1146491444.23
     Caller ID: 0
Caller ID Name: (N/A)
   DNID Digits: (N/A)
         State: Up (6)
         Rings: 0
  NativeFormat: 4
   WriteFormat: 4
    ReadFormat: 4
1st File Descriptor: 34
     Frames in: 1868
    Frames out: 7454
Time to Hangup: 0
  Elapsed Time: N/A
 Direct Bridge: Zap/21-1
Indirect Bridge: Zap/21-1
--   PBX   --
       Context: local
     Extension:
      Priority: 1
    Call Group: 2
  Pickup Group: 2
   Application: Bridged Call
          Data: Zap/21-1
   Blocking in: ast_waitfor_nandfds
     Variables:
BRIDGEPEER=Zap/21-1
DIALEDPEERNUMBER=Chrissy
SIPCALLID=759e18b264145db63faa4f835e3b9abd@10.200.20.110

Hope this helps.
-kh


By: opsys (opsys) 2006-05-01 09:03:30

!! "Just tested with asterisk-1.0.7-1," !!!

I hope you mean 1.2.7.1, What has me confused is that you CallerID Name is set to N/A.

This is from a ZAP channel to SIP. Can you do this. Can you add a wait(1) on the incomming context before you handle the call.

Second, can you do a SIP(Poly) to SIP(Snom) call and display the results.
By: khmann (khmann) 2006-05-02 14:10:54

Sorry, Asterisk-1.2.7-1 is the correct version.

The Zap channels are analog FXO's connected to an Adtran TA750 channel bank, via Digium T1 card.  CallerID is delivered standard US analog style (name + number between 1st and 2nd rings), so I don't think wait(1) will have any effect like it does on PRI, but I will check.

This issue only occurs for us with "Unavailable" PSTN callers - I reproduce the problem by dialing *67 from the PSTN to block callerID and then dial our local phone number.  Maybe there is an Asterisk flag getting set somewhere that flags the call as not having any associated CNAM data?

I will include the additional requested debugs in the next day or so.
By: Serge Vecher (serge-v) 2006-05-23 15:17:45

got debugs?
By: Kevin P. Fleming (kpfleming) 2006-05-24 16:05:30

I think there is some confusion here. You are placing outbound calls to SIP devices, so the 'callerid' setting in sip.conf should not have any effect at all. The relevant callerid information is what is coming on Zap/21 (in the case of your uploaded log file), which you didn't show us.

Please show us a 'show channel Zap/...' for one of these calls, so we can see what the CLID/CNAM on the Zap channel looks like. Thanks!
By: Serge Vecher (serge-v) 2006-05-31 09:00:06

Ok, please test with the newly released 1.2.8 and if the problem persists, please attach information requested by kpfleming. Thank you.