[Serusers] Modifying "From" header field of an INVITE message
I am relatively new to SER. My SIP service provider requires me to present the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows: From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx> They want it in E.164 format as follows: From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx> I see function only to modify the SIP URI but not the other header or SDP fields. Could someone help me with a code snippet that I can use in ser.cfg to do this conversion? Regards, SCM
Is it Level3? The From: cannot be changed. The identifier for the dialog is the From, To and CallID. None can be changed. Only the RURI (request URI) can be changed by SER. Do you have the ability to go all the way to the source and change the initial request? That would be easiest. Other solutions would require a b2bua. -g On Sat, 5 Feb 2005 11:58:05 -0800, scm-j@nuntius.com <scm-j@nuntius.com> wrote:
I am relatively new to SER. My SIP service provider requires me to present the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows:
From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx>
They want it in E.164 format as follows:
From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx>
I see function only to modify the SIP URI but not the other header or SDP fields.
Could someone help me with a code snippet that I can use in ser.cfg to do this conversion?
Regards,
SCM
_______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Greg Fausak greg@addabrand.com
Yes, it is level3. I would prefer not to change the SER source as I would need to patch in the change every time I upgrade to a new version of SER. Besides, my other SIP vendors do not require it. Would a B2BUA, such as Vovida be a good choice? Or, preferably, since I am already using Asterisk, and the INVITE is initiated by Asterisk (to SER to Level-3), is there a way to have Asterisk format the "From" in E.164 format ? Thanks.. SCM -----Original Message----- From: Greg Fausak [mailto:lgfausak@gmail.com] Sent: Saturday, February 05, 2005 3:33 PM To: scm-j@nuntius.com Cc: serusers@lists.iptel.org Subject: Re: [Serusers] Modifying "From" header field of an INVITE message Is it Level3? The From: cannot be changed. The identifier for the dialog is the From, To and CallID. None can be changed. Only the RURI (request URI) can be changed by SER. Do you have the ability to go all the way to the source and change the initial request? That would be easiest. Other solutions would require a b2bua. -g On Sat, 5 Feb 2005 11:58:05 -0800, scm-j@nuntius.com <scm-j@nuntius.com> wrote:
I am relatively new to SER. My SIP service provider requires me to
present
the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows:
From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx>
They want it in E.164 format as follows:
From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx>
I see function only to modify the SIP URI but not the other header or SDP fields.
Could someone help me with a code snippet that I can use in ser.cfg to do this conversion?
Regards,
SCM
_______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Greg Fausak greg@addabrand.com
If your providers supports Remote-Party-ID header, you can add that to the SIP body when the INVITE comes to SER. If you want to make change at Asterisk, try using 'SetCallerID' before dialing out. HTH, --- scm-j@nuntius.com wrote:
Yes, it is level3.
I would prefer not to change the SER source as I would need to patch in the change every time I upgrade to a new version of SER. Besides, my other SIP vendors do not require it.
Would a B2BUA, such as Vovida be a good choice? Or, preferably, since I am already using Asterisk, and the INVITE is initiated by Asterisk (to SER to Level-3), is there a way to have Asterisk format the "From" in E.164 format ?
Thanks..
SCM
-----Original Message----- From: Greg Fausak [mailto:lgfausak@gmail.com] Sent: Saturday, February 05, 2005 3:33 PM To: scm-j@nuntius.com Cc: serusers@lists.iptel.org Subject: Re: [Serusers] Modifying "From" header field of an INVITE message
Is it Level3?
The From: cannot be changed.
The identifier for the dialog is the From, To and CallID. None can be changed. Only the RURI (request URI) can be changed by SER.
Do you have the ability to go all the way to the source and change the initial request? That would be easiest. Other solutions would require a b2bua.
-g
On Sat, 5 Feb 2005 11:58:05 -0800, scm-j@nuntius.com <scm-j@nuntius.com> wrote:
I am relatively new to SER. My SIP service provider requires me to
present
the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows:
From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx>
They want it in E.164 format as follows:
From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx>
I see function only to modify the SIP URI but not the other header or SDP fields.
Could someone help me with a code snippet that I can use in ser.cfg to do this conversion?
Regards,
SCM
===== Girish Gopinath <gr_sh2003@yahoo.com> __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
If Asterisk is the source of the invite, by all means just change the original INVITE. I do not know how to configure Asterisk, but, I'm sure you can get the From/To formatted correctly originating from there. ---greg On Sat, 5 Feb 2005 16:46:58 -0800, scm-j@nuntius.com <scm-j@nuntius.com> wrote:
Yes, it is level3.
I would prefer not to change the SER source as I would need to patch in the change every time I upgrade to a new version of SER. Besides, my other SIP vendors do not require it.
Would a B2BUA, such as Vovida be a good choice? Or, preferably, since I am already using Asterisk, and the INVITE is initiated by Asterisk (to SER to Level-3), is there a way to have Asterisk format the "From" in E.164 format ?
Thanks..
SCM
-----Original Message----- From: Greg Fausak [mailto:lgfausak@gmail.com] Sent: Saturday, February 05, 2005 3:33 PM To: scm-j@nuntius.com Cc: serusers@lists.iptel.org Subject: Re: [Serusers] Modifying "From" header field of an INVITE message
Is it Level3?
The From: cannot be changed.
The identifier for the dialog is the From, To and CallID. None can be changed. Only the RURI (request URI) can be changed by SER.
Do you have the ability to go all the way to the source and change the initial request? That would be easiest. Other solutions would require a b2bua.
-g
On Sat, 5 Feb 2005 11:58:05 -0800, scm-j@nuntius.com <scm-j@nuntius.com> wrote:
I am relatively new to SER. My SIP service provider requires me to
present
the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows:
From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx>
They want it in E.164 format as follows:
From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx>
I see function only to modify the SIP URI but not the other header or SDP fields.
Could someone help me with a code snippet that I can use in ser.cfg to do this conversion?
Regards,
SCM
_______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Greg Fausak greg@addabrand.com
-- Greg Fausak greg@addabrand.com
Asterisk Operational Note: From Asterisk, try something similar to this: exten => _1XXXXXXXXXX.,1,SetCIDNum(+1${CALLERIDNUM}) exten => _1XXXXXXXXXX.,n,Dial(SIP/+${EXTEN}@ser.mydomain.com) This assumes you have ten digit North American caller ID digits, and that you need to hand numbers to Level3 like "+13015551212" for both ANI and DNIS. SER Development/Usage Note: While I agree that changing the "From:" is generally a bad idea, there certainly are instances in which is is a desirable thing for a SIP proxy to do. The instance below is a good example. Other examples are when the recommended privacy options are not used, so despite privacy settings elsewhere in the headers the "true" Caller ID is shown on the remote side. The only way to avoid this would be to not include the origin AOR on the outbound call. Of course, to process replies correctly, it may be necessary to mangle the headers in the inverse manner on the way back in... but that level of voodoo is left to the administrator. I agree that a b2bua is a way to do this - it may be necessary to keep state in that manner. (specific note: Asterisk _is_ a glorified b2bua.) However, it seems natural that all headers should be subject to modification if required within the proxy - not always is it required to keep state, as in this case. Is it truly the case that SER does not allow re-formatting of the From: header in any way? JT At 4:46 PM -0800 on 2/5/05, <scm-j@nuntius.com> wrote:
Yes, it is level3.
I would prefer not to change the SER source as I would need to patch in the change every time I upgrade to a new version of SER. Besides, my other SIP vendors do not require it.
Would a B2BUA, such as Vovida be a good choice? Or, preferably, since I am already using Asterisk, and the INVITE is initiated by Asterisk (to SER to Level-3), is there a way to have Asterisk format the "From" in E.164 format ?
Thanks..
SCM
-----Original Message----- From: Greg Fausak [mailto:lgfausak@gmail.com] Sent: Saturday, February 05, 2005 3:33 PM To: scm-j@nuntius.com Cc: serusers@lists.iptel.org Subject: Re: [Serusers] Modifying "From" header field of an INVITE message
Is it Level3?
The From: cannot be changed.
The identifier for the dialog is the From, To and CallID. None can be changed. Only the RURI (request URI) can be changed by SER.
Do you have the ability to go all the way to the source and change the initial request? That would be easiest. Other solutions would require a b2bua.
-g
On Sat, 5 Feb 2005 11:58:05 -0800, scm-j@nuntius.com <scm-j@nuntius.com> wrote:
I am relatively new to SER. My SIP service provider requires me to
present
the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows:
From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx>
They want it in E.164 format as follows:
From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx>
I see function only to modify the SIP URI but not the other header or SDP fields.
Could someone help me with a code snippet that I can use in ser.cfg to do this conversion?
Regards,
SCM
_______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Greg Fausak greg@addabrand.com
_______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
On 06-02 08:44, John Todd wrote:
Asterisk Operational Note: From Asterisk, try something similar to this:
exten => _1XXXXXXXXXX.,1,SetCIDNum(+1${CALLERIDNUM}) exten => _1XXXXXXXXXX.,n,Dial(SIP/+${EXTEN}@ser.mydomain.com)
This assumes you have ten digit North American caller ID digits, and that you need to hand numbers to Level3 like "+13015551212" for both ANI and DNIS.
SER Development/Usage Note: While I agree that changing the "From:" is generally a bad idea, there certainly are instances in which is is a desirable thing for a SIP proxy to do. The instance below is a good example. Other examples are when the recommended privacy options are not used, so despite privacy settings elsewhere in the headers the "true" Caller ID is shown on the remote side. The only way to avoid this would be to not include the origin AOR on the outbound call. Of course, to process replies correctly, it may be necessary to mangle the headers in the inverse manner on the way back in... but that level of voodoo is left to the administrator.
I agree that a b2bua is a way to do this - it may be necessary to keep state in that manner. (specific note: Asterisk _is_ a glorified b2bua.) However, it seems natural that all headers should be subject to modification if required within the proxy - not always is it required to keep state, as in this case. Is it truly the case that SER does not allow re-formatting of the From: header in any way?
It does require you to keep state -- you have to restore the original value on the way back, and not just in replies, but in subsequent mid-dialog requests (BYE, for example) as well. You can do it statelessly only if you know that no user agent uses the From/To URI for dialog matching. Suprisingly many implementations do, Cisco is one well-known example. Another option would be encoding the original value into the Record-Route header field generated by SER, that way SER could restore the original value from the Route header field later. It will be the user agents in this case who would keep the state. You can rewrite any header field in SER, even From/To but expect interoperability problems if you do so statelessly. Unless you are sure that all SIP implementations behave exactly as described in RFC3261, you need a B2BUA that would keep state during calls. Jan.
Hi Jan I found this response from you where you explain is possible to modify any header field in SER. I have exactly the same issue like SCM. My service provider is asking me to send a fixed user part at From URI header for all my calls for billing purpose. For example If I have 4 users registered at SER like the following user1@sipserver.mydomain.com user2@sipserver.mydomain.com user3@sipserver.mydomain.com user4@sipserver.mydomain.com I should translate all the user part before to forward their calls to the Service Provider by the following: 2102456787@sipserver.mydomain.com Where could I find information related to the command(s)/funcion(s) that make this possible? If I change the From header, what should I do to keep the original value in order to avoid loosing the call flow? Is it enough with the "Contact:" field? Regards Alberto Cruz Jan Janak wrote:
On 06-02 08:44, John Todd wrote:
Asterisk Operational Note: From Asterisk, try something similar to this:
exten => _1XXXXXXXXXX.,1,SetCIDNum(+1${CALLERIDNUM}) exten => _1XXXXXXXXXX.,n,Dial(SIP/+${EXTEN}@ser.mydomain.com)
This assumes you have ten digit North American caller ID digits, and that you need to hand numbers to Level3 like "+13015551212" for both ANI and DNIS.
SER Development/Usage Note: While I agree that changing the "From:" is generally a bad idea, there certainly are instances in which is is a desirable thing for a SIP proxy to do. The instance below is a good example. Other examples are when the recommended privacy options are not used, so despite privacy settings elsewhere in the headers the "true" Caller ID is shown on the remote side. The only way to avoid this would be to not include the origin AOR on the outbound call. Of course, to process replies correctly, it may be necessary to mangle the headers in the inverse manner on the way back in... but that level of voodoo is left to the administrator.
I agree that a b2bua is a way to do this - it may be necessary to keep state in that manner. (specific note: Asterisk _is_ a glorified b2bua.) However, it seems natural that all headers should be subject to modification if required within the proxy - not always is it required to keep state, as in this case. Is it truly the case that SER does not allow re-formatting of the From: header in any way?
It does require you to keep state -- you have to restore the original value on the way back, and not just in replies, but in subsequent mid-dialog requests (BYE, for example) as well.
You can do it statelessly only if you know that no user agent uses the From/To URI for dialog matching. Suprisingly many implementations do, Cisco is one well-known example.
Another option would be encoding the original value into the Record-Route header field generated by SER, that way SER could restore the original value from the Route header field later. It will be the user agents in this case who would keep the state.
You can rewrite any header field in SER, even From/To but expect interoperability problems if you do so statelessly. Unless you are sure that all SIP implementations behave exactly as described in RFC3261, you need a B2BUA that would keep state during calls.
Jan.
_______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi, I'm also working on this issue. I have users registered like emailusername@company.com and I want to forward this to pstn gw like telNo@company.com. As I understood this could be done by adding rpid to a user one direction and alias other direction. I'm I right? Could anyone give a link to rpid description? Alberto Cruz wrote:
Hi Jan I found this response from you where you explain is possible to modify any header field in SER.
I have exactly the same issue like SCM. My service provider is asking me to send a fixed user part at From URI header for all my calls for billing purpose. For example If I have 4 users registered at SER like the following user1@sipserver.mydomain.com user2@sipserver.mydomain.com user3@sipserver.mydomain.com user4@sipserver.mydomain.com
I should translate all the user part before to forward their calls to the Service Provider by the following: 2102456787@sipserver.mydomain.com
Where could I find information related to the command(s)/funcion(s) that make this possible?
If I change the From header, what should I do to keep the original value in order to avoid loosing the call flow? Is it enough with the "Contact:" field?
Regards
Alberto Cruz Jan Janak wrote:
On 06-02 08:44, John Todd wrote:
Asterisk Operational Note: From Asterisk, try something similar to this:
exten => _1XXXXXXXXXX.,1,SetCIDNum(+1${CALLERIDNUM}) exten => _1XXXXXXXXXX.,n,Dial(SIP/+${EXTEN}@ser.mydomain.com)
This assumes you have ten digit North American caller ID digits, and that you need to hand numbers to Level3 like "+13015551212" for both ANI and DNIS.
SER Development/Usage Note: While I agree that changing the "From:" is generally a bad idea, there certainly are instances in which is is a desirable thing for a SIP proxy to do. The instance below is a good example. Other examples are when the recommended privacy options are not used, so despite privacy settings elsewhere in the headers the "true" Caller ID is shown on the remote side. The only way to avoid this would be to not include the origin AOR on the outbound call. Of course, to process replies correctly, it may be necessary to mangle the headers in the inverse manner on the way back in... but that level of voodoo is left to the administrator.
I agree that a b2bua is a way to do this - it may be necessary to keep state in that manner. (specific note: Asterisk _is_ a glorified b2bua.) However, it seems natural that all headers should be subject to modification if required within the proxy - not always is it required to keep state, as in this case. Is it truly the case that SER does not allow re-formatting of the From: header in any way?
It does require you to keep state -- you have to restore the original value on the way back, and not just in replies, but in subsequent mid-dialog requests (BYE, for example) as well.
You can do it statelessly only if you know that no user agent uses the From/To URI for dialog matching. Suprisingly many implementations do, Cisco is one well-known example.
Another option would be encoding the original value into the Record-Route header field generated by SER, that way SER could restore the original value from the Route header field later. It will be the user agents in this case who would keep the state.
You can rewrite any header field in SER, even From/To but expect interoperability problems if you do so statelessly. Unless you are sure that all SIP implementations behave exactly as described in RFC3261, you need a B2BUA that would keep state during calls.
Jan.
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participants (7)
-
Alberto Cruz -
Girish -
Greg Fausak -
Jan Janak -
John Todd -
Mindaugas Kuprys -
scm-j＠nuntius.com