Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
Ah jeez. All I wanted to do was connect to a carrier and then perform fail over logic based on their SIP response. Not supposed to be difficult. This is what Asterisk is supposed to be good at. We have a SIP module, why not have SIP responses available to the module. Now, I have to look at the lossy HANGUPCAUSE variable and make a best guess. Not an ideal situation. We're trying to improve the ASR's we get from providers. They are low, and often they fail calls for no particular reason. They all do it, even the big ones like Verizon. Checking their responses for purpose of trying another carrier on the fly, and reporting is pretty critical. Doug. - Original Message From: Raj Jain [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 27, 2007 11:29:21 AM Subject: Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. But you already have the SIP_HEADER function, which is quite contradictory to what you say. This allows users who know what they are doing to examine headers directly. We use this a lot. What would be the harm in having a SIP_RESPONSE function or something alike? I'd agree that SIP response code should be accessible from the dial plan. Knowing the exact SIP response code could be critical for making call processing decisions. The conversion of SIP response codes to Q.931 codes (HANGUPCAUSE) is just too lossy. Building a truly protocol agnostic dial plan API is a worthy goal. But, I think it is somewhat of an unsolvable problem. The signaling protocols are very different and for various reasons people have always wanted access to native information elements carried in the protocol. Perhaps, a very simple solution for this problem could be to support a keyword such as TOPLINE in the SIP_HEADER function to fetch the topmost line in a SIP message. This will not only get the caller the response code for SIP response messages, but will also have the nice byproduct of making the Request-URI available if the message in question is a SIP request. - Raj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
Well, unfortunately for you, that is the exact opposite of the philosophy of the Asterisk codebase. Every attempt is made to genericize the channel driver interface so that you do not need to know the details of the underlying driver. Where we have failed to do so in the past, we are attempting to rectify in current approaches. The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. But you already have the SIP_HEADER function, which is quite contradictory to what you say. This allows users who know what they are doing to examine headers directly. We use this a lot. What would be the harm in having a SIP_RESPONSE function or something alike? It would allow for those who want to have this information to get it and act accordingly in the dialplan. I know I have missed this possibility, and instead tried to puzzle together information from DIAL_STATUS and HANGUP_CAUSE. I do agree with the general assumption that the dialplan should be generic, but in reality this is often not the case. You add a SIP-header to tell the client to auto-answer or to change the ring tone, or something like that. I guess that there are similiar ways customize things in IAX or ZAP, and thereby making the dialplans not so generic. Our dialplans often depend heavliy on SIP, but that is of course a result of us working in a SIP-only environment. // T ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
http://www.faqs.org/rfcs/rfc3398.html The conversion is lossy. More than 1 SIP cause code is mapped to a Q.931 cause code (in Asterisk at least). See hangup_sip2cause() in chan_sip.c True. The conversion is lossy in that respect and most of the times semantically incorrect simply because of the fundamental differences between SIP and ISUP. In fact, many new SIP response codes have been defined and will be defined in the future since RFC 3398 was written (http://www.iana.org/assignments/sip-parameters). And as far as I can tell a revision of RFC 3398 is not in works in the IETF. - Raj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Friday, October 26, 2007 11:37 AM To: Asterisk Users Subject: Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE Eric ManxPower Wieling wrote: On 10/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? There is an RFC for this. I don't know if Asterisk follows the RFC or not. http://www.faqs.org/rfcs/rfc3398.html The conversion is lossy. More than 1 SIP cause code is mapped to a Q.931 cause code (in Asterisk at least). See hangup_sip2cause() in chan_sip.c Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
On Saturday 27 October 2007 08:14:05 Torbjörn Abrahamsson wrote: Well, unfortunately for you, that is the exact opposite of the philosophy of the Asterisk codebase. Every attempt is made to genericize the channel driver interface so that you do not need to know the details of the underlying driver. Where we have failed to do so in the past, we are attempting to rectify in current approaches. The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. But you already have the SIP_HEADER function, which is quite contradictory If you read all of what I said, you'd see that it's not contradictory at all. I said that there are places where we've failed to maintain that segregation and that we're working to rectify that, where possible. to what you say. This allows users who know what they are doing to examine headers directly. We use this a lot. What would be the harm in having a SIP_RESPONSE function or something alike? It would allow for those who want to have this information to get it and act accordingly in the dialplan. I know I have missed this possibility, and instead tried to puzzle together information from DIAL_STATUS and HANGUP_CAUSE. I do agree with the general assumption that the dialplan should be generic, but in reality this is often not the case. You add a SIP-header to tell the client to auto-answer or to change the ring tone, or something like that. I guess that there are similiar ways customize things in IAX or ZAP, and thereby making the dialplans not so generic. Our dialplans often depend heavliy on SIP, but that is of course a result of us working in a SIP-only environment. Not all of us are working in a SIP-only environment. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. But you already have the SIP_HEADER function, which is quite contradictory to what you say. This allows users who know what they are doing to examine headers directly. We use this a lot. What would be the harm in having a SIP_RESPONSE function or something alike? I'd agree that SIP response code should be accessible from the dial plan. Knowing the exact SIP response code could be critical for making call processing decisions. The conversion of SIP response codes to Q.931 codes (HANGUPCAUSE) is just too lossy. Building a truly protocol agnostic dial plan API is a worthy goal. But, I think it is somewhat of an unsolvable problem. The signaling protocols are very different and for various reasons people have always wanted access to native information elements carried in the protocol. Perhaps, a very simple solution for this problem could be to support a keyword such as TOPLINE in the SIP_HEADER function to fetch the topmost line in a SIP message. This will not only get the caller the response code for SIP response messages, but will also have the nice byproduct of making the Request-URI available if the message in question is a SIP request. - Raj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
I think you can use the 'ngrep' command to see the sip packets coming in using the sip listening port. I dont know the exact command though, you will have to lookit up urself. you will see the sip packets coming into ur system and in those packets you can see the response code. On 10/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
Rizwan Hisham wrote: I think you can use the 'ngrep' command to see the sip packets coming in using the sip listening port. I dont know the exact command though, you will have to lookit up urself. you will see the sip packets coming into ur system and in those packets you can see the response code. On 10/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? There is an RFC for this. I don't know if Asterisk follows the RFC or not. http://www.faqs.org/rfcs/rfc3398.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
Eric ManxPower Wieling wrote: On 10/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? There is an RFC for this. I don't know if Asterisk follows the RFC or not. http://www.faqs.org/rfcs/rfc3398.html The conversion is lossy. More than 1 SIP cause code is mapped to a Q.931 cause code (in Asterisk at least). See hangup_sip2cause() in chan_sip.c Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
Thanks. I am quite familiar with ngrep. I was asking how I could get the SIP response code from the dial plan. Doug. - Original Message From: Rizwan Hisham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 26, 2007 6:18:50 AM Subject: Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE I think you can use the 'ngrep' command to see the sip packets coming in using the sip listening port. I dont know the exact command though, you will have to lookit up urself. you will see the sip packets coming into ur system and in those packets you can see the response code. On 10/25/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
On Friday 26 October 2007 16:13:11 Douglas Garstang wrote: Thanks. I am quite familiar with ngrep. I was asking how I could get the SIP response code from the dial plan. You cannot. And you shouldn't have to. The dialplan should be generic to all protocols, not customized to SIP. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
Tilghman Lesher wrote: On Friday 26 October 2007 16:13:11 Douglas Garstang wrote: Thanks. I am quite familiar with ngrep. I was asking how I could get the SIP response code from the dial plan. You cannot. And you shouldn't have to. The dialplan should be generic to all protocols, not customized to SIP. I disagree that is SHOULDN'T be available. We should have the option to test for a SIP error return code of a specific type if we want. It can make the logic for SIP calls considerably nicer -- especially in a complex AGI. Generally, I've found that dialplans should NOT be generic. They should be customised to your particular environment. If your environment is SIP, then the customisation should be available. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
On Friday 26 October 2007 19:47:27 SIP wrote: Tilghman Lesher wrote: On Friday 26 October 2007 16:13:11 Douglas Garstang wrote: Thanks. I am quite familiar with ngrep. I was asking how I could get the SIP response code from the dial plan. You cannot. And you shouldn't have to. The dialplan should be generic to all protocols, not customized to SIP. I disagree that is SHOULDN'T be available. We should have the option to test for a SIP error return code of a specific type if we want. It can make the logic for SIP calls considerably nicer -- especially in a complex AGI. Generally, I've found that dialplans should NOT be generic. They should be customised to your particular environment. If your environment is SIP, then the customisation should be available. Well, unfortunately for you, that is the exact opposite of the philosophy of the Asterisk codebase. Every attempt is made to genericize the channel driver interface so that you do not need to know the details of the underlying driver. Where we have failed to do so in the past, we are attempting to rectify in current approaches. The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE
SIP wrote: Tilghman Lesher wrote: On Friday 26 October 2007 16:13:11 Douglas Garstang wrote: I was asking how I could get the SIP response code from the dial plan. You cannot. And you shouldn't have to. The dialplan should be generic to all protocols, not customized to SIP. That's the reasoning someone always comes up with. Me too. :) And it's true for probably 90 % of the people who are asking for SIP response codes to be available in the dialplan. I disagree that is SHOULDN'T be available. We should have the option to test for a SIP error return code of a specific type if we want. It can make the logic for SIP calls considerably nicer -- especially in a complex AGI. Generally, I've found that dialplans should NOT be generic. They should be customised to your particular environment. If your environment is SIP, then the customisation should be available. I agree. Dialplans should be as generic as possible but if you need/want to customize any of the default behavior and if know what you're doing Asterisk shouldn't stop you. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting SIP Response Code from HANGUPCAUSE
I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
