XMLCDR Question: <callflow profile_index="x">
I am experimenting with XML CDRs and I was curious about this part of the record. It seems that with each transfer to a different extension I'm getting a new callflow profile. That's good - I just want to make sure that my understanding of this is correct. Will a new callflow profile get generated for each successive transfer to a different extension? Also, are there other scenarios that generate a new callflow profile? Lastly, is there a way to force a new callflow profile to occur from within the dialplan? (That last one is just a curiosity, although it would help me handle some quirky CDR needs with the IVR that I'm working on.) Thanks! -MC P.S. - as per usual, any info provided here will go straight to the wiki!
Hi , I created the following in /conf/directory/default <gateways> <gateway name="inphonex"> <param name="username" value="3434141"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="605431"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> then the following dialplan in /conf/dialplan/extensions <extension name="3434141"> <condition field="destination_number" expression="^(3434141)$"> <action application="bridge" data="sofia/gateway/inphonex/$1"/> </condition> </extension> I am getting 2008-07-09 11:48:26 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3434141@192.168.0.102] can anybody tell what I am missing ? thanks Jair Santos
There are two ways to add gateways. If you added this in the directory as is thats wrong. http://wiki.freeswitch.org/wiki/Voicepulse.xml That page has an example of one in the directory and one in the sip profile. This would be correct XML for the directory case. <include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3434141"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="605431"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="605431"/> </params> </user> </include> /b On Jul 9, 2008, at 1:59 PM, Jair Santos wrote:
Hi ,
I created the following in /conf/directory/default
<gateways> <gateway name="inphonex"> <param name="username" value="3434141"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="605431"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways>
then the following dialplan in /conf/dialplan/extensions <extension name="3434141"> <condition field="destination_number" expression="^(3434141)$"> <action application="bridge" data="sofia/gateway/inphonex/$1"/> </condition> </extension>
I am getting
2008-07-09 11:48:26 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3434141@192.168.0.102]
can anybody tell what I am missing ?
thanks
Jair Santos _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
Thank you Brian, I did as you said changing the file on conf/directory/default but I am still getting 2008-07-09 12:48:26 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3434141@192.168.0.102] Any other ideas/clues ? regards Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 12:21 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings
There are two ways to add gateways. If you added this in the directory as is thats wrong.
http://wiki.freeswitch.org/wiki/Voicepulse.xml
That page has an example of one in the directory and one in the sip profile.
This would be correct XML for the directory case. <include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3434141"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="605431"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="605431"/> </params> </user> </include> /b
On Jul 9, 2008, at 1:59 PM, Jair Santos wrote:
Hi ,
I created the following in /conf/directory/default
<gateways> <gateway name="inphonex"> <param name="username" value="3434141"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="605431"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways>
then the following dialplan in /conf/dialplan/extensions <extension name="3434141"> <condition field="destination_number" expression="^(3434141)$"> <action application="bridge" data="sofia/gateway/inphonex/$1"/> </condition> </extension>
I am getting
2008-07-09 11:48:26 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3434141@192.168.0.102]
can anybody tell what I am missing ?
thanks
Jair Santos _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Brian West sip:brian@freeswitch.org
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Is this an inbound call on a registered gateway? /b On Jul 9, 2008, at 2:53 PM, Jair Santos wrote:
Thank you Brian,
I did as you said changing the file on conf/directory/default but I am still getting
2008-07-09 12:48:26 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3434141@192.168.0.102]
Any other ideas/clues ?
regards
Jair Santos
Brian West sip:brian@freeswitch.org
It is a registered gateway. Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 12:55 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings Is this an inbound call on a registered gateway? /b On Jul 9, 2008, at 2:53 PM, Jair Santos wrote: Thank you Brian, I did as you said changing the file on conf/directory/default but I am still getting 2008-07-09 12:48:26 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3434141@192.168.0.102] Any other ideas/clues ? regards Jair Santos Brian West sip:brian@freeswitch.org
Sorry, I am trying an outbond call -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jair Santos Sent: Wednesday, July 09, 2008 1:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings It is a registered gateway. Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 12:55 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings Is this an inbound call on a registered gateway? /b On Jul 9, 2008, at 2:53 PM, Jair Santos wrote: Thank you Brian, I did as you said changing the file on conf/directory/default but I am still getting 2008-07-09 12:48:26 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [3434141@192.168.0.102] Any other ideas/clues ? regards Jair Santos Brian West sip:brian@freeswitch.org
You're looking at the wrong side of this now. Do you have a user 3434141 in domain 192.168.0.102 ? The output of sofia status will help. On Jul 9, 2008, at 3:04 PM, Jair Santos wrote:
Sorry,
I am trying an outbond call
Brian West sip:brian@freeswitch.org
Yes I do. This user is configured on an ATA connected to same network The config is like. SIP Server Address (IP or URL) Outbound Proxy (IP address or URL,if any) SIP User ID Account ID Authentication Password (For security,password does not display) Please don't mind about the real server address and user id . They are correct. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 1:07 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings You're looking at the wrong side of this now. Do you have a user 3434141 in domain 192.168.0.102 ? The output of sofia status will help. On Jul 9, 2008, at 3:04 PM, Jair Santos wrote: Sorry, I am trying an outbond call Brian West sip:brian@freeswitch.org
API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia@192.168.1.117:5060 RUNNING (0) nat profile sip:mod_sofia@24.67.78.200:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@24.67.78.200:5080 RUNNING (0) inphonex gateway sip:3462101@sip.varphonex.com REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED ============================================================================ ===================== -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 1:05 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings What is the output of "sofia status" On Jul 9, 2008, at 3:01 PM, Jair Santos wrote: It is a registered gateway. Jair Santos Brian West sip:brian@freeswitch.org
Your domain is 192.168.1.117 check the config on your phone correctly and remove the 192.168.0.102 and replace it with the proper value. If this is an SPA phone make sure to click Advanced Admin On Jul 9, 2008, at 3:07 PM, Jair Santos wrote:
API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================== internal profile sip:mod_sofia@192.168.1.117:5060 RUNNING (0) nat profile sip:mod_sofia@24.67.78.200:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@24.67.78.200:5080 RUNNING (0) inphonex gateway sip:3462101@sip.varphonex.com REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 1:05 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings
What is the output of "sofia status"
On Jul 9, 2008, at 3:01 PM, Jair Santos wrote:
It is a registered gateway.
Jair Santos
Brian West sip:brian@freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
I am sorry that I passed the wrong information about the IP address and userid. I can assure that they are correct and the ATA is properly configured. Please see below. API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia@192.168.1.117:5060 RUNNING (0) nat profile sip:mod_sofia@24.67.78.200:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@24.67.78.200:5080 RUNNING (0) inphonex gateway sip:3462101@sip.varphonex.com REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED ============================================================================ ===================== And the ATA SIP Server Address (IP or URL) Outbound Proxy (IP address or URL,if any) SIP User ID Account ID Authentication Password (For security,password does not display) Display Name (Optional) Use DNS SRV Yes No User ID is phone number Yes No SIP Registration Yes No Unregister On Reboot Yes No NAT Traversal (STUN) Yes No Register Expiration Minutes (Default is 1 hour, max 45 days) Local SIP Port (Default 5060) Local RTP Port (1024-65535, default The user 3462101 is a valid inphonex user.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 1:15 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings
Your domain is 192.168.1.117 check the config on your phone correctly and remove the 192.168.0.102 and replace it with the proper value. If this is an SPA phone make sure to click Advanced Admin
On Jul 9, 2008, at 3:07 PM, Jair Santos wrote:
API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
internal profile
sip:mod_sofia@192.168.1.117:5060
RUNNING (0) nat profile
sip:mod_sofia@24.67.78.200:5070
RUNNING (0) default alias internal ALIASED external profile
sip:mod_sofia@24.67.78.200:5080
RUNNING (0) inphonex gateway
sip:3462101@sip.varphonex.com
REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 1:05 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings
What is the output of "sofia status"
On Jul 9, 2008, at 3:01 PM, Jair Santos wrote:
It is a registered gateway.
Jair Santos
Brian West sip:brian@freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
Brian West sip:brian@freeswitch.org
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I just checked inphonex and FreeSWITCH-mod_sofia/1.0.0 is registered in their server. 3462101 sip.varphonex.com sip:3462101-g471esoujgor0@10.0.5.66:5060;useradd=24.67.78.200;userport=5080; transport=udp 2008-07-09 17:21:51 FreeSWITCH-mod_sofia/1.0.0-exported No -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jair Santos Sent: Wednesday, July 09, 2008 1:41 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings I am sorry that I passed the wrong information about the IP address and userid. I can assure that they are correct and the ATA is properly configured. Please see below. API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia@192.168.1.117:5060 RUNNING (0) nat profile sip:mod_sofia@24.67.78.200:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@24.67.78.200:5080 RUNNING (0) inphonex gateway sip:3462101@sip.varphonex.com REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED ============================================================================ ===================== And the ATA SIP Server Address (IP or URL) Outbound Proxy (IP address or URL,if any) SIP User ID Account ID Authentication Password (For security,password does not display) Display Name (Optional) Use DNS SRV Yes No User ID is phone number Yes No SIP Registration Yes No Unregister On Reboot Yes No NAT Traversal (STUN) Yes No Register Expiration Minutes (Default is 1 hour, max 45 days) Local SIP Port (Default 5060) Local RTP Port (1024-65535, default The user 3462101 is a valid inphonex user.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 1:15 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings
Your domain is 192.168.1.117 check the config on your phone correctly and remove the 192.168.0.102 and replace it with the proper value. If this is an SPA phone make sure to click Advanced Admin
On Jul 9, 2008, at 3:07 PM, Jair Santos wrote:
API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
internal profile
sip:mod_sofia@192.168.1.117:5060
RUNNING (0) nat profile
sip:mod_sofia@24.67.78.200:5070
RUNNING (0) default alias internal ALIASED external profile
sip:mod_sofia@24.67.78.200:5080
RUNNING (0) inphonex gateway
sip:3462101@sip.varphonex.com
REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Wednesday, July 09, 2008 1:05 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Initial testings
What is the output of "sofia status"
On Jul 9, 2008, at 3:01 PM, Jair Santos wrote:
It is a registered gateway.
Jair Santos
Brian West sip:brian@freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users
Brian West sip:brian@freeswitch.org
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every time a new caller profile is installed into a channel it will stack another callflow so each time it returns to the dialplan which is caused mainly by the transfer app or the originate FSAPI when it does an outbound call then transfers it to an extension. On Wed, Jul 9, 2008 at 1:53 PM, Michael Collins <mcollins@fcnetwork.com> wrote:
I am experimenting with XML CDRs and I was curious about this part of the record. It seems that with each transfer to a different extension I'm getting a new callflow profile. That's good – I just want to make sure that my understanding of this is correct.
Will a new callflow profile get generated for each successive transfer to a different extension? Also, are there other scenarios that generate a new callflow profile? Lastly, is there a way to force a new callflow profile to occur from within the dialplan? (That last one is just a curiosity, although it would help me handle some quirky CDR needs with the IVR that I'm working on.)
Thanks!
-MC
P.S. – as per usual, any info provided here will go straight to the wiki!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Is there a way to tell what caused that new caller profile to happen? If it was a transfer or if it was an originate from a FSAPI? On Wed, Jul 9, 2008 at 3:10 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
every time a new caller profile is installed into a channel it will stack another callflow so each time it returns to the dialplan which is caused mainly by the transfer app or the originate FSAPI when it does an outbound call then transfers it to an extension.
On Wed, Jul 9, 2008 at 1:53 PM, Michael Collins <mcollins@fcnetwork.com> wrote:
I am experimenting with XML CDRs and I was curious about this part of the record. It seems that with each transfer to a different extension I'm getting a new callflow profile. That's good – I just want to make sure that my understanding of this is correct.
Will a new callflow profile get generated for each successive transfer to a different extension? Also, are there other scenarios that generate a new callflow profile? Lastly, is there a way to force a new callflow profile to occur from within the dialplan? (That last one is just a curiosity, although it would help me handle some quirky CDR needs with the IVR that I'm working on.)
Thanks!
-MC
P.S. – as per usual, any info provided here will go straight to the wiki!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- --Joe "Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy)
both cases are a transfer On Wed, Jul 9, 2008 at 2:33 PM, Joseph Bajin <josephbajin@gmail.com> wrote:
Is there a way to tell what caused that new caller profile to happen? If it was a transfer or if it was an originate from a FSAPI?
On Wed, Jul 9, 2008 at 3:10 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
every time a new caller profile is installed into a channel it will stack another callflow so each time it returns to the dialplan which is caused mainly by the transfer app or the originate FSAPI when it does an outbound call then transfers it to an extension.
On Wed, Jul 9, 2008 at 1:53 PM, Michael Collins <mcollins@fcnetwork.com> wrote:
I am experimenting with XML CDRs and I was curious about this part of the record. It seems that with each transfer to a different extension I'm getting a new callflow profile. That's good – I just want to make sure that my understanding of this is correct.
Will a new callflow profile get generated for each successive transfer to a different extension? Also, are there other scenarios that generate a new callflow profile? Lastly, is there a way to force a new callflow profile to occur from within the dialplan? (That last one is just a curiosity, although it would help me handle some quirky CDR needs with the IVR that I'm working on.)
Thanks!
-MC
P.S. – as per usual, any info provided here will go straight to the wiki!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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"Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy) _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
But if you didn't tell us that these were transfers, how would we know is my question? On Wed, Jul 9, 2008 at 3:43 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
both cases are a transfer
On Wed, Jul 9, 2008 at 2:33 PM, Joseph Bajin <josephbajin@gmail.com> wrote:
Is there a way to tell what caused that new caller profile to happen? If it was a transfer or if it was an originate from a FSAPI?
On Wed, Jul 9, 2008 at 3:10 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
every time a new caller profile is installed into a channel it will stack another callflow so each time it returns to the dialplan which is caused mainly by the transfer app or the originate FSAPI when it does an outbound call then transfers it to an extension.
On Wed, Jul 9, 2008 at 1:53 PM, Michael Collins <mcollins@fcnetwork.com> wrote:
I am experimenting with XML CDRs and I was curious about this part of the record. It seems that with each transfer to a different extension I'm getting a new callflow profile. That's good – I just want to make sure that my understanding of this is correct.
Will a new callflow profile get generated for each successive transfer to a different extension? Also, are there other scenarios that generate a new callflow profile? Lastly, is there a way to force a new callflow profile to occur from within the dialplan? (That last one is just a curiosity, although it would help me handle some quirky CDR needs with the IVR that I'm working on.)
Thanks!
-MC
P.S. – as per usual, any info provided here will go straight to the wiki!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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-- --Joe
"Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy) _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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-- --Joe "Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy)
every <callflow> is an instance of a caller_profile. every channel has one to start out with and a new one for every time it's transferred. so if there is more than one it's been transferred one way or another. On Wed, Jul 9, 2008 at 3:23 PM, Joseph Bajin <josephbajin@gmail.com> wrote:
But if you didn't tell us that these were transfers, how would we know is my question?
On Wed, Jul 9, 2008 at 3:43 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
both cases are a transfer
On Wed, Jul 9, 2008 at 2:33 PM, Joseph Bajin <josephbajin@gmail.com> wrote:
Is there a way to tell what caused that new caller profile to happen? If it was a transfer or if it was an originate from a FSAPI?
On Wed, Jul 9, 2008 at 3:10 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
every time a new caller profile is installed into a channel it will stack another callflow so each time it returns to the dialplan which is caused mainly by the transfer app or the originate FSAPI when it does an outbound call then transfers it to an extension.
On Wed, Jul 9, 2008 at 1:53 PM, Michael Collins <mcollins@fcnetwork.com> wrote:
I am experimenting with XML CDRs and I was curious about this part of the record. It seems that with each transfer to a different extension I'm getting a new callflow profile. That's good – I just want to make sure that my understanding of this is correct.
Will a new callflow profile get generated for each successive transfer to a different extension? Also, are there other scenarios that generate a new callflow profile? Lastly, is there a way to force a new callflow profile to occur from within the dialplan? (That last one is just a curiosity, although it would help me handle some quirky CDR needs with the IVR that I'm working on.)
Thanks!
-MC
P.S. – as per usual, any info provided here will go straight to the wiki!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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-- --Joe
"Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy) _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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-- --Joe
"Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy)
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
also they are stacked in reverse order so the most recent one is first. On Wed, Jul 9, 2008 at 3:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
every <callflow> is an instance of a caller_profile.
every channel has one to start out with and a new one for every time it's transferred. so if there is more than one it's been transferred one way or another.
On Wed, Jul 9, 2008 at 3:23 PM, Joseph Bajin <josephbajin@gmail.com> wrote:
But if you didn't tell us that these were transfers, how would we know is my question?
On Wed, Jul 9, 2008 at 3:43 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
both cases are a transfer
On Wed, Jul 9, 2008 at 2:33 PM, Joseph Bajin <josephbajin@gmail.com> wrote:
Is there a way to tell what caused that new caller profile to happen? If it was a transfer or if it was an originate from a FSAPI?
On Wed, Jul 9, 2008 at 3:10 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
every time a new caller profile is installed into a channel it will stack another callflow so each time it returns to the dialplan which is caused mainly by the transfer app or the originate FSAPI when it does an outbound call then transfers it to an extension.
On Wed, Jul 9, 2008 at 1:53 PM, Michael Collins < mcollins@fcnetwork.com> wrote:
I am experimenting with XML CDRs and I was curious about this part of the record. It seems that with each transfer to a different extension I'm getting a new callflow profile. That's good – I just want to make sure that my understanding of this is correct.
Will a new callflow profile get generated for each successive transfer to a different extension? Also, are there other scenarios that generate a new callflow profile? Lastly, is there a way to force a new callflow profile to occur from within the dialplan? (That last one is just a curiosity, although it would help me handle some quirky CDR needs with the IVR that I'm working on.)
Thanks!
-MC
P.S. – as per usual, any info provided here will go straight to the wiki!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com<MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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-- --Joe
"Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy) _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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-- --Joe
"Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy)
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
In my experience the first profile_index corresponds to the originate command and subsequent indices correspond to the various transfers. Here's an example from a real cdr (redacted) that has four indices. Note that they are in reverse order, so the most recent profile_index appears first in the xml file: <callflow dialplan="XML" profile_index="4"> <extension name="tod_example" number="4913"> ...snip... </extension> <caller_profile> <username></username> <dialplan>XML</dialplan> <caller_id_name>FreeSWITCH</caller_id_name> <ani></ani> <aniii></aniii> <caller_id_number>8005551212</caller_id_number> <network_addr></network_addr> <rdnis>person</rdnis> <destination_number>4913</destination_number> <uuid>7181f6d2-95dd-42ee-a589-544f575ebf9a</uuid> <source>src/switch_ivr_originate.c</source> <context>default</context> <chan_name>OpenZAP/1:1/5595551212</chan_name> <originatee> <originatee_caller_profile> <username></username> <dialplan>XML</dialplan> <caller_id_name>FreeSWITCH</caller_id_name> <ani></ani> <aniii></aniii> <caller_id_number></caller_id_number> <network_addr></network_addr> <rdnis>person</rdnis> <destination_number>4/a/1160</destination_number> <uuid>fee6d853-44c4-43cf-999f-32a5579e129c</uuid> <source>src/switch_ivr_originate.c</source> <context>default</context> <chan_name>OpenZAP/4:1/1160</chan_name> </originatee_caller_profile> </originatee> </caller_profile> <times> <created_time>1215550893566509</created_time> <profile_created_time>1215550957250480</profile_created_time> <progress_time>1215550893942396</progress_time> <progress_media_time>1215550895995423</progress_media_time> <answered_time>1215550904242447</answered_time> <hangup_time>1215550973271729</hangup_time> <transfer_time>0</transfer_time> </times> </callflow> <callflow dialplan="XML" profile_index="3"> <extension name="person" number="person" current_app="hangup"> ...snip... </extension> <caller_profile> <username></username> <dialplan>XML</dialplan> <caller_id_name>FreeSWITCH</caller_id_name> <ani></ani> <aniii></aniii> <caller_id_number>8005551212</caller_id_number> <network_addr></network_addr> <rdnis>825595551212</rdnis> <destination_number>person</destination_number> <uuid>7181f6d2-95dd-42ee-a589-544f575ebf9a</uuid> <source>src/switch_ivr_originate.c</source> <context>default</context> <chan_name>OpenZAP/1:1/5595551212</chan_name> </caller_profile> <times> <created_time>1215550893566509</created_time> <profile_created_time>1215550905873831</profile_created_time> <progress_time>1215550893942396</progress_time> <progress_media_time>1215550895995423</progress_media_time> <answered_time>1215550904242447</answered_time> <hangup_time>0</hangup_time> <transfer_time>1215550957250480</transfer_time> </times> </callflow> <callflow dialplan="XML" profile_index="2"> <extension name="tod_example" number="825595551212"> <application app_name="set" app_data="open=true"></application> ...snip... </extension> <caller_profile> <username></username> <dialplan>XML</dialplan> <caller_id_name>FreeSWITCH</caller_id_name> <ani></ani> <aniii></aniii> <caller_id_number>8005409011</caller_id_number> <network_addr></network_addr> <rdnis>1/a/5595551212</rdnis> <destination_number>825595551212</destination_number> <uuid>7181f6d2-95dd-42ee-a589-544f575ebf9a</uuid> <source>src/switch_ivr_originate.c</source> <context>default</context> <chan_name>OpenZAP/1:1/5595551212</chan_name> </caller_profile> <times> <created_time>1215550893566509</created_time> <profile_created_time>1215550895997527</profile_created_time> <progress_time>1215550893942396</progress_time> <progress_media_time>1215550895995423</progress_media_time> <answered_time>1215550904242447</answered_time> <hangup_time>0</hangup_time> <transfer_time>1215550905873831</transfer_time> </times> </callflow> <callflow profile_index="1"> <caller_profile> <username></username> <dialplan></dialplan> <caller_id_name>FreeSWITCH</caller_id_name> <ani></ani> <aniii></aniii> <caller_id_number>8005551212</caller_id_number> <network_addr></network_addr> <rdnis></rdnis> <destination_number>1/a/5595551212</destination_number> <uuid>7181f6d2-95dd-42ee-a589-544f575ebf9a</uuid> <source>src/switch_ivr_originate.c</source> <context>default</context> <chan_name>OpenZAP/1:1/5595551212</chan_name> </caller_profile> <times> <created_time>1215550893566509</created_time> <profile_created_time>1215550893566509</profile_created_time> <progress_time>1215550893942396</progress_time> <progress_media_time>1215550895995423</progress_media_time> <answered_time>0</answered_time> <hangup_time>0</hangup_time> <transfer_time>1215550895997527</transfer_time> </times> </callflow> </cdr> Notice how profile_index="1" doesn't have the extension tag, nor does it have a 'dialplan="XML"' property. In my setup this is the result of the originate command, in this case: originate openzap/1/a/5595551212 825595551212 A successful originate places the call into my "tod_example" extension (see profile_index="2") where it gets processed. It gets transferred a few more times which is why there are other profile_index values. BTW, an unsuccessful originate will have just the one profile_index unless "continue_on_fail=true". (Someone please confirm that - I don't use continue_on_fail at all.) If I have a USER_BUSY or UNALLOCATED result for my originate cmd then I simply have an xmlcdr with a single profile_index, plus the usual stuff like the <variables>. If the originate is successful then I have the various profile_index entries depending upon what happens to the call in my dialplan. For example, I don't ignore early media, so I get "successful" originate commands when I get ringing after I dial. I have a waitforanswer.js script that waits 30 secs and if no answer then it sets a channel variable and hangs up. Hope this helps... -MC ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, July 09, 2008 12:43 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] XMLCDR Question: <callflow profile_index="x"> both cases are a transfer On Wed, Jul 9, 2008 at 2:33 PM, Joseph Bajin <josephbajin@gmail.com> wrote: Is there a way to tell what caused that new caller profile to happen? If it was a transfer or if it was an originate from a FSAPI? On Wed, Jul 9, 2008 at 3:10 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: every time a new caller profile is installed into a channel it will stack another callflow so each time it returns to the dialplan which is caused mainly by the transfer app or the originate FSAPI when it does an outbound call then transfers it to an extension. On Wed, Jul 9, 2008 at 1:53 PM, Michael Collins <mcollins@fcnetwork.com> wrote: I am experimenting with XML CDRs and I was curious about this part of the record. It seems that with each transfer to a different extension I'm getting a new callflow profile. That's good - I just want to make sure that my understanding of this is correct. Will a new callflow profile get generated for each successive transfer to a different extension? Also, are there other scenarios that generate a new callflow profile? Lastly, is there a way to force a new callflow profile to occur from within the dialplan? (That last one is just a curiosity, although it would help me handle some quirky CDR needs with the IVR that I'm working on.) Thanks! -MC P.S. - as per usual, any info provided here will go straight to the wiki! _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --Joe "Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy) _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (5)
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Anthony Minessale
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Brian West
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Jair Santos
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Joseph Bajin
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Michael Collins