Asterisk is really pissing me off.
Can someone tell me why this doesn't cause SRV lookups to be done on
outbound calls:
[general]
srvlookup=yes
...
[proxy]
type=peer
host=pstn.voip.com
insecure=very
context=test
qualify=yes
exten => s,2,Dial(SIP/${EXTEN}@proxy,20,rt)
NO SRV LOOKUP!
While the following DOES cause an SRV lookup to be done...
exten => s,2,Dial(SIP/${EXTEN}@pstn.voip.com,20,rt)
So... if I put the domain directly into the dial command an SRV lookup is done.
If I reference it in sip.conf, an SRV lookup *IS NOT* done. WTF???
Dear Douglas!> Asterisk is really pissing me off. > Can someone tell me why this doesn't cause SRV lookups to be done on > outbound calls:In general: If you are missing documentation then you are warmly invited to write and enhance the existing one (e.g. the Wiki) wherever you see fit. In particular you might want to edit this and add what you (and me) have learned: http://www.voip-info.org/wiki/view/Asterisk+SIP+srvlookup> So... if I put the domain directly into the dial command an SRV lookup > is done. If I reference it in sip.conf, an SRV lookup *IS NOT* done. > WTF???I did some work for you and searched bugs.digium.com for "srvlookup". And look at what I found: http://bugs.digium.com/view.php?id=1805 http://bugs.digium.com/view.php?id=2081 Reading the bug notes you'll find that this is known - and probably even intended - behaviour. If you dislike it: File a bug report, write a patch, or find someone that's going to write it for you. Asterisk is an open source project, remember? Cheers, Philipp
Phillip. The link to the Wiki is woefully indadequate. I have no problem adding to the documentation, as soon as I bloody understand it myself. The two bug links you provided appear to be almost completely unrelated to what I asked about except they touch on the subject of SRV lookups. If you can't reference the proxy to dial in sip.conf, then you lose the ability to set a whole bunch of options (such as qualify which is required for detecting CONGESTION when the proxy is down etc). If I stick with an IP/host and refer to what's in sip.conf, Asterisk ends URI's like sip:user@192.168.10.7 which is but ugly and breaks SIP in general. Oh, and you know what, why is it assumed that to use open source software I have to be a seasoned C programmer who can contribute to the code? Where is that requirement stipulated? At this point I'm almost ready to throw Asterisk out the window and suggest we spend $300,000 on the Sylantro solution. Doug. -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: Sunday, December 11, 2005 10:13 AM To: Douglas Garstang; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SRV Lookups *ARRGH!* Dear Douglas!> Asterisk is really pissing me off. > Can someone tell me why this doesn't cause SRV lookups to be done on > outbound calls:In general: If you are missing documentation then you are warmly invited to write and enhance the existing one (e.g. the Wiki) wherever you see fit. In particular you might want to edit this and add what you (and me) have learned: http://www.voip-info.org/wiki/view/Asterisk+SIP+srvlookup> So... if I put the domain directly into the dial command an SRV lookup > is done. If I reference it in sip.conf, an SRV lookup *IS NOT* done. > WTF???I did some work for you and searched bugs.digium.com for "srvlookup". And look at what I found: http://bugs.digium.com/view.php?id=1805 http://bugs.digium.com/view.php?id=2081 Reading the bug notes you'll find that this is known - and probably even intended - behaviour. If you dislike it: File a bug report, write a patch, or find someone that's going to write it for you. Asterisk is an open source project, remember? Cheers, Philipp
Hey, That is not what I meant!!!! I L O V E ASTERISK, every other PBX I have had to deal with, always had some limitation, I am only using 1.0.7 and really have found nothing limiting. We run ISDN 30 line, Reception get 440+ calls per day, dial out 23,000 calls per month all fully integrated into an old PABX. My reference was spouse to come across "Go back to old style PBX and you will be disappointed!" My public apologies for anyone on the list. Thanks James -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of trixter aka Bret McDanel Sent: Monday, 12 December 2005 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Limitations On Mon, 2005-12-12 at 09:00 +1000, James Sturges wrote:> Or..... > > You can go back to a Traditional PBX and really experience the meaning of > the phrase { significant "limitations". } >That is a really bad excuse for limitations however, and actually does more harm than good. While it may be true that asterisk has fewer limitations than another product to say that your option is to use asterisk or something else more limiting doesnt get any of the problems fixed. At least the person you replied to gave constructive answers to remove some of the limitations, such as looking at CVS/bugtracker for patches or paying money to get someone motivitated to fix them. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group