Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve
Hey steve I remember a tip somewhere where they used a conference room and added all the users into that conference muted, then kicked them out at the end of the call. Sorry I can't remember at all where this was but it looked like it could work. How did you get the autoanswer to work, I have tried different patches and non work? joel -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Steve Clark Sent: Friday, May 20, 2005 9:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] paging thru sipura-841 Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005
Hello , I want some tips guidance i am sure this topic discuss alot in list,i try my best to solve it by myself try googling looking wiki everywhere but no luck question is iax-iax trunking not working setting,trying each n every option server2 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [saim] username=saim secret=saim type=friend host=dynamic context=from-sip disallow=all allow=gsm [noman] username=saim secret=noman type=friend host=dynamic context=from-sip disallow=all allow=gsm [asteriskser1] type=friend ;auth=md5 ;secret=qwerty context=local ;host=dynamic defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no server1 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [user1] username=user1 secret=user1 type=friend host=dynamic context=from-sip disallow=all allow=gsm [user2] username=user2 secret=user2 type=friend host=dynamic context=from-sip disallow=all allow=gsm [test2] type=friend context=local defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no I am using Kiax soft phone on both servers using codec GSM asterisk latest stable version OS SLES9 ,any help is highly appreciated i had look almost every place in wiki regarding iax trunking but all in vein. Thanks In Advance.
Okay sounds like a stupid question but just to be clear do you have some sort of timer on both machines? Joel -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Adnan Ahmed Sent: Saturday, May 21, 2005 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX-IAX Trunking not works Hello , I want some tips guidance i am sure this topic discuss alot in list,i try my best to solve it by myself try googling looking wiki everywhere but no luck question is iax-iax trunking not working setting,trying each n every option server2 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [saim] username=saim secret=saim type=friend host=dynamic context=from-sip disallow=all allow=gsm [noman] username=saim secret=noman type=friend host=dynamic context=from-sip disallow=all allow=gsm [asteriskser1] type=friend ;auth=md5 ;secret=qwerty context=local ;host=dynamic defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no server1 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [user1] username=user1 secret=user1 type=friend host=dynamic context=from-sip disallow=all allow=gsm [user2] username=user2 secret=user2 type=friend host=dynamic context=from-sip disallow=all allow=gsm [test2] type=friend context=local defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no I am using Kiax soft phone on both servers using codec GSM asterisk latest stable version OS SLES9 ,any help is highly appreciated i had look almost every place in wiki regarding iax trunking but all in vein. Thanks In Advance. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005
Joel Duffield wrote:> Okay sounds like a stupid question but just to be clear do you have some > sort of timer on both machines? > > JoelAnd of course you would want more than one channel to see the benefits of trunking. -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)