similar to: inbound SIP funnies

Hi All, I was made aware today that some of my calls coming in are not going to voicemail... Below are some logs, and the macro that should run on the incoming_pstn context for that extension. I can see that theres a non-zero exit before it gets to voicemail, but I've no idea why. In this case theres 2 SIP clients to sim-call. On other occasions it works fine. In the CDR logs, I can see

We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to Dahdi (2.1.0.4). Everything seemed to go smooth with the exception of meetme. Meetme seems to not be able to find a zap channel for conferencing. We use voice introductions in our conference bridge and it seems to break that feature. The error from the console is.... # app_meetme.c:2593 find_conf: No Zap channel available for

Hi, Running 1.4.15. I've a SIP user as below. My default context in sip.conf is [incomming_pstn] I'm having trouble with inbound calls going to the wrong context. [test-ubi] username=test-ubi type=friend secret=XXXXXXX host=dynamic canreinvite=no context=testinbound nat=yes allow=ulaw allow=gsm allow=alaw qualify=no the testinbound context includes the code to

Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0

Hi, I have an FXO card connected to my phone line which works in Asterisk as Zap/1. Is there any way of detecting whether something else is on the line before picking up on this channel? For example, I dont want to pick up and dial out on the line if someone is on it using another phone (which is connected directly to the line, rather than through Asterisk). Also, when an incoming call comes

Hi All, I've got an MD3200 modem which is working as a Zaptel FXO interface for Asterisk (X100P clone I believe). It seems to work, but on incoming or outgoing calls I can hear the other party ok but when I speak, I hear my voice echo back at me (quite quietly but it's distracting!) on everything I say. The other party doesn't seem to hear the echo - they just say I sound a bit quiet.

Hi All, I'm going to be upgrading our 1.2 Asterisk system. At the moment we use the Enicomms SLN files. Are there major differences in the 1.4 default voicefile packs, or will I be able to re-use Enicomms?? In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? Thanks Adrian

Hi, I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it into production. Ive done this by installing 1.4.18 onto the VM, putting my config files in place and then installing 1.4.37 over the top (which is what I'd have to do on production). I've found a few issues in the config files, but nothing I couldn't handle until... I hit inbound IAX issues. My

Hi, I tried to use Early Media: exten => 1,1,Playback(demo-thanks,noanswer) same => n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using "Progress()" in the first priority. I tried "pri set debug on span 1" and got the

Good Day list, Anyone know if there is a way to have the AgentCallBackLogin function play a voice file after the agent picks up the phone? If this is not an available feature, any ideas on the difficulty in making this feature? Example: Extensions.conf exten?=>?700,1,AgentCallbackLogin(${CALLERIDNUM}|?AnnounceCAllQue-TechSu pport?); ....... exten => s,6,Queue(queue1)

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but <10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems

Hello, I wanted to switch from using Dialogic/Eicon cards to using Digium's T-1 cards. When I purchased a sample card the salesperson assured me there was documentation specific to the DAHDI interface. Now that I'm digging in, I'm finding it's documented a lot like Linux -- one must read the fairly uncommented source code. I don't have a problem with this generally, but here

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Script started on Tue Mar 3 07:35:34 2015 doctor.nl2k.ab.ca//usr/source/openssh-SNAP-20150304$ make tests [ -d `pwd`/regress ] || mkdir -p `pwd`/regress [ -d `pwd`/regress/unittests ] || mkdir -p `pwd`/regress/unittests [ -d `pwd`/regress/unittests/test_helper ] || mkdir -p `pwd`/regress/unittests/test_helper [ -d `pwd`/regress/unittests/sshbuf ] || mkdir -p `pwd`/regress/unittests/sshbuf [ -d

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Hello, all. We have successfully connected our new Asterisk 1.6.1.1 PBX to Vitelity's network and have been very happy with them thus far. However, we'd like to use domains in our sip.conf to facilitate routing in our multi-tenant environment. We also like to set allowexternaldomains=no for security. However, this breaks our inbound PSTN calling from Vitelity. Is it possible to use

Hi every body I have a linux server with Intel(R) Xeon(TM) CPU 3.20GHz , and 2 Gigabit of RAM , kernel version 2.6.22.6 , and 2 Intel 82541PI Gigabit Ethernet controllers In simple situation i would like to limit bandwidth for 2 customers 1) ( to 34 Mb/s ) and 2) 68 Mb/s . My conf is as below ///////////////////////////////////////////////////// #IFACE FACONG THE CUSTOMERS /sbin/tc