similar to: Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?

Hi Guys, I have tried every single rule I could into iptables but I can't register this VPS to a provider Spikko. Finally I did an iptable accept on INPUT, OUTPUT, and FORWARD, for ports 0:65000 just to test things and still I can't register to the provider. I can easily register to another provider which uses port 5060 and I can make calls. But sip show registry only show Request Sent

Hi there, I want to set up an extension set that acts different depending on the count of used lines. I have a EuroISDN E1 board with mISDN and I only want to offer 10 lines. Therefore I set up a global variables LINES in the general section of extensions.conf and instantiate it with 0. I a call is incoming I check the LINES variable wether is 10 or more. If so I make a call transfer. If not

Hello list members, We're trying to get MWI notifications on our ATA device and we set it to send SUBSCRIBE messages to Asterisk, but it gets UNAUTHORIZED messages, despite the fact that we set the following lines in its settings in sip.conf: subscribemwi=yes mailbox=21 at from-extensions We need help in understanding how this works and what we are doing wrong. This is the SIP debug

hello I am using ser with asterisk asterisk on 5070 (on back end) ser on 5060 (on front end) i am getting all requests at asterisk. i tried by changing asterisk port bindport=5090 but still getting all requests from sjphone at asterisk. can any one tell what is the reason regrads Kamran __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on

Hello to all, Is there an "how-to" for asterisk and setting up a t38 fax gateway (SIP) ? I look at http://bugs.digium.com/view.php?id=5090 to patch asterisk chan_sip.c file. What are the next steps to get a t38 fax gateway with asterisk ? Regards Harry PS: I use hylafax server. ___________________________________________________________________________ Faites de

Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in

Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a

Hi list, I think, I found a bug in the replication setup, that drops (custom) flags like Thunderbird labels. I have set up a simple 2 node setup to reproduce and explain it: Host adove.mpi-klsb.mpg.de and bdove.mpi-klsb.mpg.de (doveconf -n attached) have dovecot-imapd (from repo.dovecot.org) installed on an ext4 filesystem. They only have one account "test" via /etc/dovecot/userdb

Hi, I've got report about issue when dovecot fails to start and there is no error logged (error goes only to stderr) situation: 1) dovecot is running 2) dovecot is automatically updated to new version (by yum update daemon), after update, dovecot is restarted (it's part of update script) 3) new dovecot fails to start (for whatever reason) result: dovecot not running and no error

Trying to copy the (binary) header of a input file directly to an output file, I've had repeatable seg faults. The call: writeChar(hdr, outfh, nchars=6144) when hdr just contains one empty string seems to be the culprit. The stack traces weren't all that illuminating, with sig 11 in memory-related functions following this. But in src/main/connections.c it looks like do_writechar

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

Hi list, this question was already posted a few years ago (https://www.dovecot.org/list/dovecot/2014-November/098585.html). I already asked the original queriest and he told me, that he never got an solution or workaround but it was not important enough for him. When using replication in conjunction with sieve vacations, the .dovecot.lda-dupes file is not synced with the other server. So when

is there recomended source files for t.38 pass? latest cvs does not work for me. is it possible publish working src? ------------------------------------ turby -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060214/8dfb088c/attachment.htm

I would like to use smbmount on a SGI running IRIX 6.5.1m. The distribution of samba for IRIX does not include the executables smbmount or smbmnt. Is there a source for these executables? Regards, Fred -------------------------------------- Fred Kunzler Phone: (281) 575-5090 91-1SE-42C Pager: (800) 618-5062 10200 Bellaire Blvd. Fax: (281) 575-5473 Houston, TX 77072 E-mail:

For some reason, I can't get call pickup to work between Sip phones or between Sip and Zap phones. All phones are in the same call group and pickup group (1). The source code was downloaded and built as of today 11/15/03. Here's what's in sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=aliens ; ; SIP Entry for sipura line 1 ; This

Hello, I know that Mog was trying to get the bug-tracker cleaned up as the number of bugs has increased substantially over the past few months. I figured that I would do my part to bring attention to a couple of bugs that are interesting and have some wide reaching impact. That being said: http://bugs.digium.com/view.php?id=5970 - This has a patch created by Corydon, but it has segfaulted on

(This is a shameless copy-paste from the note I posted on http://bugs.digium.com/view.php?id=5090) I have again backported the whole T.38 shebang to the stable branch. The port was based on two versions of the t38passthrough branch: r19125, the latest unconflicted automerge, and r13623, the latest version without the new chan_sip flag structure. Basically, the port contains everything that

Thanks for these informations I would have prefer to receive them from asterisk-users instead of serusers !! May be they are sleeping . Ok i have not installed spandsp because of i don't find some scripts like in hylafax for mail2fax fax2mail i've just patched chan_sip.c Regards Harry --- Alexandr Dubovikov <shurik@start4.info> a ?crit : > On Wed, Apr 26, 2006 at 11:27:01AM

Hi, I'm connecting Asterisk v. 1.4.10 to Zanzibar Open IVR that acts as a SIP trunk. Since recognition didn't work correctly, I've troubleshot with Wireshark and saw that RTP stream is first send to one port on SIP trunk and then when first RTP packet arrives in opposite direction (from TTS part of Zanzibar - it's a prompt) Asterisk starts sending to the same RTP port -

Hi list, hi devs, On 26.11.18 16:26, Patrick Cernko wrote: > Hi list, > > I think, I found a bug in the replication setup, that drops (custom) > flags like Thunderbird labels. I have set up a simple 2 node setup to > reproduce and explain it: > > Host adove.mpi-klsb.mpg.de and bdove.mpi-klsb.mpg.de (doveconf -n > attached) have dovecot-imapd (from repo.dovecot.org)