similar to: Trunk is not released

Is there an up and running provider of SIP termination in Brazil? I know that there are some people building on a SIP termination solution. But who as it up and running ? Best regards, Han -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040921/f1043e19/attachment.htm

Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

<font face="Default Sans Serif,Verdana,Arial,Helvetica,sans-serif" size="2"> <div>Thanks Aki,</div><div><br></div><div>do you know if this list is read by dovecot developers team too? Could you point me to the right list otherwise?</div> <div><br><br>Raffaele

Has anyone seen good scripts or documentation on Voicemail groups? We are looking to have a system where you can send a voicemail to multiple mailboxes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060518/777a7b83/attachment.htm

asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Board member of Connectech Greater Detroit www.connectech.org ________________________________ Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph.

-- *Larry "Yako" Junior* *:On Air Analyst/personality :Online Radio Broadcaster "YakoRadio" **:V.O. Artist* *:Image/Com Producer* *:Audio Engineer :Media Talent** **facebook.com/yakoradionetworks <http://www.facebook.com/yakoradionetworks> twitter.com/yakoradio myspace.com/yakoenturadio yakoradio.com youtube.com/yakoenturadio soundcloud.com/yakoradio

Hi all, I would like to understand the meaning of "Stale mailbox lock file detected" In index-storage.c (https://mis.dopa.go.th/atmail/server_source/dovecot/src/lib-storage/index/index-storage.c) there is this notice "Stale mailbox lock file detected" ... I have a java application which connects via javamail to a dovecot server and I have some anomalies when that message

Spinning off from another topic...what are people using for QoS / Shaping? I'm using Wondershaper script with OK results...but I'd like better. Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091008/96b14a3e/attachment.htm

my guess would be that the sphynx user does not have permission to write to the folder /var/log/icecast try running the following command: chown sphynx.sphynx /var/log/icecast then as root you can execute: icecast -c /usr/local/etc/icecast.xml The reason you can run the icecast command as root is because when you set <chroot>1</chroot> in the <security> section (which you

I'm filtering that is being deliverd to postfix mail server with amavisd-new . I want spam with spam f level 1 - 8 to ad a tag any everything above to be delete is this posebol? If yes how? Met vriendelijk groet, Bas van Dikkenberg GISkit bv BFVD1-RIPE Tel: +3130-6340430 Fax: +3130-6342433 Prive Tel: +3130-6372769 Mob:

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

Does anyone know how I can append to different user recorded voice files within a dial plan? For example Asterisk ask caller a question and records the answer, then ask another question record the answer to the end of the first answer - so when it's played back, all the answers are in one playback. TIA Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Please let me know your interest in following. Location: Columbia, SC Duration: 12 months+ Rate: $65/hr 1099/c2c Required Skills: RUBY, RAILS, GIT, MYSQL, CUCUMBER, RSPEC, JQUERY, EXCELLENT ORAL AND WRITTEN COMMUNICATION SKILLS, TEST-DRIVEN DEVELOPMENT, LINUX, OS X, JSON, COMMAND LINE, SQL, SSH, HAML, SCSS Thanks Sandeep Sandeep Jain Software People Inc. www.softwarepeople.us

I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered

Bruce et al. I'm posting a new thread with the "Top Posting" subject so I won't draw complaints about "hijacking" the 4-port thread. Top Posting refers to the practice of sending a message with a reply at the top and including the entire thread below the reply. I prefer this. If I'm actively following a thread, the most-recent information appears at the top

Hi, i want to keep track of user mailbox size, I'm using doveadm mailbox status -u <user> vsize INBOX for a given mailbox this is under reporting ( by a lot) is this recursive? what shoud i be using to get an accurate disk usage metric? thanks Paul. -- ------------------------- Paul Macdonald IFDNRG Ltd Web and video hosting ------------------------- t: 0131 5548070 m:

I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184

On 09/10/2020 12:52, lists wrote: > I have to say I'm totally baffled since I do nothing when LetsEncrypt renews the certificate. > > I know the cert has been updated because the mail clients asks me if I trust the certificate. Curious. The mail clients really shouldn't ask anything when encountering a valid certificate. Are you sure the client isn't asking you to trust an

Hi, I am looking for voip termination all over the world especially based on IAX or SIP. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040124/25e16736/attachment.htm