Earlier today, I was in the chat room trying to get help with a problem where one of my extensions didn't ring back when calling it. Any other extension would dial that extension, 305, and just hear dead air. 305 would actually ring, but when picked up, no sound was on the line. This is what the log looked like when trying to call 305: http://pastebin.freeswitch.org/5119 Here is the pastebin that shows a few of my extensions: http://pastebin.freeswitch.org/5126 You can see that extension 305 is registered with the caller id name Testing & Accountability. That is what I set in the caller id field on the Aastra 9143i phone. I finally figured out that if I remove the ampersand, the problem goes away. My question is... Is this a limitation (bug?) of the Aastra firmware, sofia, sip in general, or something in freeswitch? Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x301
That is correct. I further verified it was the ampersand causing the problem by setting the Caller ID field to a shorter string, like B & H. With that, I still see the same no ringback problem.
On 8/15/2008 at 2:20 PM, Brian West <brian@freeswitch.org> wrote: Ok without the '&' its fine.. and with its not?
/b On Aug 15, 2008, at 1:16 PM, Ben Holtsclaw wrote: My question is... Is this a limitation (bug?) of the Aastra firmware, sofia, sip in general, or something in freeswitch? Brian West sip:brian@freeswitch.org
Are you setting it in the phone or in the freeswitch directory? On Aug 15, 2008, at 1:29 PM, Ben Holtsclaw wrote:
That is correct. I further verified it was the ampersand causing the problem by setting the Caller ID field to a shorter string, like B & H. With that, I still see the same no ringback problem.
Brian West sip:brian@freeswitch.org
I had it set in the freeswitch directory to the employee's first name and last name (no ampersand). <variable name="effective_caller_id_name" value="Ellis Ayers"/> <variable name="effective_caller_id_number" value="305"/> The tech that installed the phone set the Caller ID field on the phone itself to Testing & Accountability. That is the name that I was seeing registered with sofia in freeswitch.
On 8/15/2008 at 2:39 PM, Brian West <brian@freeswitch.org> wrote: Are you setting it in the phone or in the freeswitch directory?
On Aug 15, 2008, at 1:29 PM, Ben Holtsclaw wrote: That is correct. I further verified it was the ampersand causing the problem by setting the Caller ID field to a shorter string, like B & H. With that, I still see the same no ringback problem. Brian West sip:brian@freeswitch.org
Give me the output of "sofia status profile internal" for when it has the & in it. /b On Aug 15, 2008, at 2:01 PM, Ben Holtsclaw wrote:
I had it set in the freeswitch directory to the employee's first name and last name (no ampersand).
<variable name="effective_caller_id_name" value="Ellis Ayers"/> <variable name="effective_caller_id_number" value="305"/>
The tech that installed the phone set the Caller ID field on the phone itself to Testing & Accountability. That is the name that I was seeing registered with sofia in freeswitch.
Brian West sip:brian@freeswitch.org
The sofia status output is attached. I also attached what I currently have for 305 in the directory folder.
On 8/15/2008 at 3:15 PM, Brian West <brian@freeswitch.org> wrote: Give me the output of "sofia status profile internal" for when it has the & in it.
/b On Aug 15, 2008, at 2:01 PM, Ben Holtsclaw wrote: I had it set in the freeswitch directory to the employee's first name and last name (no ampersand). <variable name="effective_caller_id_name" value="Ellis Ayers"/> <variable name="effective_caller_id_number" value="305"/> The tech that installed the phone set the Caller ID field on the phone itself to Testing & Accountability. That is the name that I was seeing registered with sofia in freeswitch. Brian West sip:brian@freeswitch.org
Ampersand is not a valid token in an RFC 3261 display name. Perhaps that is part of the problem. Within the SIP message the display name must be enclosed within quotes if it contains certain characters. I ran into to this with ANI from an FXO. Twinkle phone, as an example, will reject the message. On Friday 15 August 2008, Ben Holtsclaw wrote:
The sofia status output is attached. I also attached what I currently have for 305 in the directory folder.
On 8/15/2008 at 3:15 PM, Brian West <brian@freeswitch.org> wrote:
Give me the output of "sofia status profile internal" for when it has the & in it.
/b
On Aug 15, 2008, at 2:01 PM, Ben Holtsclaw wrote:
I had it set in the freeswitch directory to the employee's first name and last name (no ampersand).
<variable name="effective_caller_id_name" value="Ellis Ayers"/> <variable name="effective_caller_id_number" value="305"/>
The tech that installed the phone set the Caller ID field on the phone itself to Testing & Accountability. That is the name that I was seeing registered with sofia in freeswitch.
Brian West
sip:brian@freeswitch.org
So the Aastra doesn't encode it before sending it? /b On Aug 15, 2008, at 3:46 PM, Robert Dyck wrote:
Ampersand is not a valid token in an RFC 3261 display name. Perhaps that is part of the problem. Within the SIP message the display name must be enclosed within quotes if it contains certain characters. I ran into to this with ANI from an FXO. Twinkle phone, as an example, will reject the message.
Brian West sip:brian@freeswitch.org
In my example the call came from an spa3000 PSTN gateway that was set up to forward to a SIP user. The spa3000 sent the display name without quotes. On Friday 15 August 2008, Brian West wrote:
So the Aastra doesn't encode it before sending it?
/b
On Aug 15, 2008, at 3:46 PM, Robert Dyck wrote:
Ampersand is not a valid token in an RFC 3261 display name. Perhaps that is part of the problem. Within the SIP message the display name must be enclosed within quotes if it contains certain characters. I ran into to this with ANI from an FXO. Twinkle phone, as an example, will reject the message.
Brian West sip:brian@freeswitch.org
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participants (3)
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Ben Holtsclaw
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Brian West
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Robert Dyck