Hey Girish,
Thanks for swift reply!
I had come to same conclusion about dialplan too... however, I am not sure the best way of proceeding. In our network, most users will have both alphanumeric usernames and 4-digit traditional style extension numbers - both for those with 12-key phones and for easy mapping to external numbers. For this reason, it seemed easiest to base the Asterisk VM boxes on those 4-digit extension numbers... then rather than add an extension for every user in *, I just have a single rule using the {exten} variable. Otherwise I need some way of mapping alphanumeric and digit codes to the same mailbox in *... very confusing... I thought that it would be a lot easier to let * handle the simplest one, and make Ser always pass it the format it's expecting... there must be way to do a lookup(alias) and then re-write the URI with the alias before it gets forwarded.?
Sorry if my explanation about the 404 is confusing... it's confusing for me too! Attached is the ser.cfg (I hope it retains the indentation!)
Many thanks again,
Dave
-----Original Message----- From: GR S [mailto:gr_sh2003@yahoo.com] Sent: 28 July 2004 22:04 To: Dave Bath Cc: serusers@lists.iptel.org Subject: RE: [Serusers] Asterisks to ser to asterisk (voicemail)
Hello,
--- Dave Bath dave@fuuz.com wrote:
<snip>
(1) If a user is called with their alphanumeric ID instead of their numerical alias, * does not pick up the call. This is as expected, as the dial pattern in * is _[1-9] [0-9] [0-9] [0-9]. However, it must
be
fairly common to call people with their email addresses for example... so how can I make ser pass the alias to * instead of the alpha URI?
I think the problem is in your dialplan. Try adding something like this: exten => _.,1,Dial(SIP/....
(2) If a user is offline, I get a 404 immediately, instead of anything else - for example diverting immediately to vm. I don't quite understand this at the moment.. as I have the t_on_failure set up
before
the location lookups... does the t_on_failure not catch 404 failures?
Sorry, I dont understand this problem. Looks to me as a misconfiguration in the config file. Please post your ser.cfg. There are many in this forum who can tell you about the problem after verifying the config file. I saw your previous post and excerpts from the config file, but did not quite understand that.
Thanks again everyone,
Dave
Best Regards,
===== Girish Gopinath gr_sh2003@yahoo.com
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Howdy,
I've got a situation where I need to be able to forward a call. Fairly typical:
CISCOGATEWAY->SERPROXY->UAC
Best case would be to honor the 302 Moved Temporarily from the UAC. I can't figure out how to make it happen.
I've seen some talk about cpl, which seems like it could be used to forward a call. Has anyone used cpl to forward calls?
I use Asterisk, and I know that using this scenario:
CISCOGATEWAY->SERPROXY->ASTERISK->UAC
I can bounce the call with a 302 and ASTERISK performs a b2bua and starts another outbound call, ie:
CISCOGATEWAY->SERPROXY->ASTERISKB2BUA->SERPROXY->CISCOGATEWAY
Are there any other ways of getting a call forwarded from/to a gateway?
---greg
Greg Fausak www.AddaBrand.com (US) 469-546-1265
At 01:46 AM 7/29/2004, Greg Fausak wrote:
Howdy,
I've got a situation where I need to be able to forward a call. Fairly typical:
CISCOGATEWAY->SERPROXY->UAC
Best case would be to honor the 302 Moved Temporarily from the UAC. I can't figure out how to make it happen.
I don't understand what the problem is... if you wish the UA to send 302, then configure it to do so, SER will relay it, the gateway will initiate another attempt if permitted to do so.
I've seen some talk about cpl, which seems like it could be used to forward a call. Has anyone used cpl to forward calls?
If you are referring to things such as personalized user call forwarding on busy, on unavailable, etc., yes, that's what CPL is good for and we actually used it this way.
-jiri
I use Asterisk, and I know that using this scenario:
CISCOGATEWAY->SERPROXY->ASTERISK->UAC
I can bounce the call with a 302 and ASTERISK performs a b2bua and starts another outbound call, ie:
CISCOGATEWAY->SERPROXY->ASTERISKB2BUA->SERPROXY->CISCOGATEWAY
Are there any other ways of getting a call forwarded from/to a gateway?
-jiri
Jiri Kuthan jiri@iptel.org wrote: [...]
I've seen some talk about cpl, which seems like it could be used to forward a call. Has anyone used cpl to forward calls?
If you are referring to things such as personalized user call forwarding on busy, on unavailable, etc., yes, that's what CPL is good for and we actually used it this way.
What if I just want to forward all missed calls (forward on busy, on no answer, etc) to a ser/sems server? Do I need to use CPL for that? Every example I've seen for forwarding to a voicemail server, where a failure_route is used, sends the call to a specific mailbox (at least as far as I understand it). For example, from the end of http://www.mit.edu/afs/athena/project/sip/sip.edu/ser.shtml:
failure_route[4] { append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); }
I've seen many other examples that do essentially the same thing. I want each user to have their own voicemail box (or, rather, their voicemail should be emailed to their own registered email address). It seems to me this is the way the voicemail system was designed to work, and for calls that go straight to voicemail (e.g. the user isn't online) it works fine.
But for some reason routing to the voicemail server via forward() or t_relay() from a failure_route gets me a fast busy instead. The odd thing is that the call is being forwarded to the VM server, but the caller never hears the announcement and just gets that fast busy immediately. Can someone please point me in the right direction to sort this one out?
Thanks.
At 03:24 AM 8/7/2004, O'Shaughnessy Evans wrote:
Jiri Kuthan jiri@iptel.org wrote: [...]
I've seen some talk about cpl, which seems like it could be used to forward a call. Has anyone used cpl to forward calls?
If you are referring to things such as personalized user call forwarding on busy, on unavailable, etc., yes, that's what CPL is good for and we actually used it this way.
What if I just want to forward all missed calls (forward on busy, on no answer, etc) to a ser/sems server? Do I need to use CPL for that? Every example I've seen for forwarding to a voicemail server, where a failure_route is used, sends the call to a specific mailbox (at least as far as I understand it). For example, from the end of http://www.mit.edu/afs/athena/project/sip/sip.edu/ser.shtml:
failure_route[4] { append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); }
I've seen many other examples that do essentially the same thing. I want each user to have their own voicemail box (or, rather, their voicemail should be emailed to their own registered email address). It seems to me this is the way the voicemail system was designed to work, and for calls that go straight to voicemail (e.g. the user isn't online) it works fine.
Indeed -- you can do it using SER configuration for global (as opposed to personal) preferences.
But for some reason routing to the voicemail server via forward() or t_relay() from a failure_route gets me a fast busy instead. The odd thing is that the call is being forwarded to the VM server, but the caller never hears the announcement and just gets that fast busy immediately. Can someone please point me in the right direction to sort this one out?
You need to troubleshoot config file and message dumps -- I can't say more without seeing them either. One thing which strikes me is use of forward -- for all the "do-something-on-error", you need stateful processing, i.e., use the t_.* action family.
-jiri
Dave,
--- Dave Bath dave@fuuz.com wrote:
Hey Girish,
<snip>
Ser always pass it the format it's expecting... there must be way to do a lookup(alias) and then re-write the URI with the alias before it gets forwarded.?
I am sorry. I haven't used 'aliases' and MSILO module. So my knowledge is very limited on using these features. We also have a setup somewhat similer to what you had described, but we have a plug-in for SER from which we fetch the records from the database, modify the uri and send it to Asterisk. This works fine for us and as you said, the ${EXTEN} in Asterisk dialplan handles all such numbers.
Sorry if my explanation about the 404 is confusing... it's confusing for me too! Attached is the ser.cfg (I hope it retains the indentation!)
From your ser.cfg:
if ((method=="INVITE" || method=="ACK") && t_newtran() ) { t_reply("404", "Not Found");
I think this is why you get the 404 message failure route. Instead try this:
if ((method == .... { t_on_failure("4"); t_relay(); break; }
Dave
Best Regards,
===== Girish Gopinath gr_sh2003@yahoo.com
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Dave Bath -
GR S -
Greg Fausak -
Jiri Kuthan -
O'Shaughnessy Evans