similar to: Call hangs up after exactly 1 minute

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Hey list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A --> Z. In my sip.conf : register => user1:passwd1 at server/user1 register => user2:passwd2 at server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer

Hi, how do I solve this in Rails: Lets say I have a User and a Message. Now, a Message is sent from one User to another User, which means that in the Message I have to reference a User twice! This seems to be a problem, if I do not user one-to-many or many-to- many relationships, because it seems to me as if I could reference a Model of the same type only once. However, I would like the Message

Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of

Hi All; Asterisk version is: 1.8.5.0 But I see at the consol the following warning and really I did google but did not understand if it is bug or related to settings: [Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468) for sip_reinvite_retry for dialog 3c581fa96f2b-53yysntgjmwb in handle_response_invite But actually, we see some SNOM IP

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward the call to

I'm getting the following warning with SOME calls on 1.8.14.1 Is it a cause for concern? Is there a way to fix it? I can't tell for sure if it is impacting calls or not. WARNING[27796]: chan_sip.c:20497 handle_response_invite: just did sched_add waitid(4077) for sip_reinvite_retry for dialog 358b7af85cf24cf0609cb0195b273935@[ip removed] in handle_response_invite

We are testing the new T38 passthrough code (SVN-oej-t38passthrough-r13347): - we are using a Sipura SPA-2100 as the T.38 user device - we are using a Patton SmartNode 2400 as the T.38/PRI gateway - we are using Asterisk in the middle We have the following in the [general] section of our sip.conf: t38pt_udptl = yes t38pt_rtp = yes When a fax call comes in from the SmartNode to Asterisk

Greetings, i've been posted a message to this list in july, which had one response. Thanks for that idea! Unfortunately asterisk is only a hobby, and did not have much time dealing with the problem since. My original letter was long, i wouldn't post it again, the archive url is http://archives.free.net.ph/message/20070710.053008.c02209c0.en.html Since than i've upgraded to

Hello all, I've been using * for quite some time and yesterday I decided to add sipbroker to my config. It was pretty simple and it works for some numbers (e.g. I can call *258-9123, UK date & time - which is on the "phone numbers you can call" page -) but fails for some others. For example I've got a friend who's at freephonie so to call him, I would dial

-uzzi PS: If you're not seeing any connection information, be sure to double-check the IP address is correct. Learned that lesson the hard way =\ On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg <jr at amanue.com> wrote: > Let's say I have two Asterisk boxes, A and B. I am trying to get A to do > SIP registration on B, so an extension for A can dial SIP phones covered by >

I'm running dovecot --version 2.3.11.3 (502c39af9) I'm setting up folder sharing. Following https://wiki.dovecot.org/SharedMailboxes/Shared I've configured a folder to be shared, but it's not seen/accessible in the target user's Mail folder tree. My config includes, mail_plugins = virtual acl protocol imap { mail_plugins = $mail_plugins imap_acl imap_quota

Hello list, using asterisk 1.4.30 I have the strangest problem that some SIP accounts can register to my Asterisk and others not. I see no connection between all those that can register or all those that can't. It's not a firewall problem as all register to port 5060 and the range 5060 --> 5064 is open. It's just very strange that some can register and other not. Any

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:55:28 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > > > > > On

Hi all, I have noticed a strange behavior with dovecot (tested with 2.2.13). I use shared folders to share mailboxes between users. I have a namespace called "Accounts" that hosts the shared folder for the users (prefix Accounts/%%n/). However, When I issue an IMAP select command on a random non-existent mailbox name under "Accounts", dovecot auto-creates it and pollutes my

Thank you Rowland - really clear example and explanation. >From your example, this is what I would see, once the RFC2307 attributes had been added: root at testdc2:~# getent passwd user2 user2:*:3000015:10000:Jane Doe:/home/SAMBADOM/user2:/bin/false root at testdc2:~# net cache flush root at testdc2:~# getent passwd user2 user2:*:10004:10000:Jane Doe:/home/SAMBADOM/user2:/bin/false [ ... wait

How can I set the prefered codec between 2 calling parties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI shows : [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to