similar to: Faulty voicemail

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

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3659 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061121/10d6b6f8/smime-0001.bin

Hi All, Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server, and I see "Got SIP response 405 "Method Not Allowed" back from 192.168.3.64" but the phone seems to work ok. Any ideas where it falls over in the SIP protocol? I've included this in the debug below. ubiphone*CLI> <-- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines) Nat

I am trying to page my Grandstream GXP-2000 phones and keep getting the error message: Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete destination '' supplied. How can I fix this error? The two contexts below do either one-way paging or two-way paging to all Grandstream phones in a list. [One_Way_Page_GROUP] ; one to many page exten =>

Hi, I've a few working asterisk servers, all seeing the same symptom, but they are all based on the same configs. A SIP inbound INVITE message is coming in to an extension (not a peer) eg 555 at ourserver.com A tcpdump clearly shows the INVITE coming in, but asterisk seems to be ignoring it (theres no reply outbound packet). All the source/dest IPs and ports look good. A

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

Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for

Hi all, I'm trying to get a better understanding of the 'Hint' priority for use with Snom phones, etc. >From the Wiki, I understand that the following will work: exten => 200,hint,sip/200 exten => 200,1,Dial(sip/200) exten => 201,hint,sip/300 exten => 201,1,Dial(sip/201) When exactly is 'Hint' executed? After someone dials '200' or

Hello all, One of the buildings I have an asterisk box deployed in is used by two small companies on two floors. They have an agreement between them whereby they'll answer each other's incoming calls and take messages if the office is empty / everyone is on the phone. Each of them has an ISDN BRI delivered to asterisk via zaphfc, then dropped into a context as follows: exten =>

Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn'

I've got this in extensions.conf: [macro-stdexten] exten => s,1,Dial(${ARG2},30,p) exten => 6015555555,1,Macro(stdexten,200,SIP/200&SIP/201&SIP/203&SIP/${VOICEPULSE_GATEWAY_OUT_A}/+15045555555) Where the real numbers have been replaced with 5555555. What I'm trying to do is ring my cell phone in addition to the local extensions. Funny thing is the cell phone rings

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've got Asterisk SVN-trunk-r9059 currently running on Fedora Core 4 w/ 2 eyebeam softphones and 2 Grandstream GXP-2000. At my desk I've got the grandstream and the GXP-2000 I would like to ring both. Using macro-exten-vm and dialparties.agi Macro(exten-vm,200,200-202) the caller is sent to the unavailable voicemail but if I use

I have googled this to death, and all I get are reference to the MoH needing a Zaptel timing source, and then people saying no they don't any more. -- Set Response Timeout to 2 -- Executing BackGround("SIP/211-57ba", "my-greeting") in new stack -- Playing 'my-greeting' (language 'en') == CDR updated on SIP/211-57ba -- Executing

I'm trying to set a variable in one context and use it in another: [c1] MY_GROUP = SIP/200&SIP/201 [c2] include c1 exten s,1,NoOp(${MY_GROUP}) The noop prints out a blank string. In [c1], I've tried exten s,1,SetVar(MY_GROUP=SIP/200&SIP/201) and exten s,1,SetVar(_MY_GROUP=SIP/200&SIP/201) and exten s,1,SetVar(__MY_GROUP=SIP/200&SIP/201) and exten

*Hello group! * *Now I?m trying to solve following problem. I have a requirement that each employee should have **SIP phone at home, SIP phone in office, cell phone with same user. * *I want all those 3 phones to be ?one extension?. So, if someone calls our company number and dials my extension - I?d like 3 phones to ring at the same time.* *e.g. Extension 555 for all the places and when

Hi, Having seen the messages on the dev list a couple of weeks ago about chan_sebi I thought I would try to get it going on my system. I am using 1.6.1.4 so I first upgraded the driver to work with 1.6. I think it is mostly correct and the bit I have wrong shouldn't be causing me the problems I am seeing. If your are interested my chan_sebi 1.4 to 1.6 patch is at

Hi there, I have a Polycom phone that has two extensions registered to it, let's say 200 & 201. Is there anyway to code in the Asterisk dialplan to show BOTH lines are busy when either of 200 or 201 are in use? Reason is that my Polycom phones will show the presence info via BLF red light, but I would have to have 2 separate entries in the monitoring phone (one for each extension), even

Dear list, I am trying to learn how to use Zap-things in Asterisk. While loading Asterisk verbosely I get this error: [chan_zap.so]Warning, flexibel rate not heavily tested! => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Mar 12 17:19:01 WARNING[5563]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or directory Mar 12

Hi all, I've been using the following method to limit calls to sip clients to 1: exten => 200,1,SetGroup(200) exten => 200,2,CheckGroup(1) exten => 200,3,Dial(SIP/200) exten => 200,103,Busy This works fine for a single extension. However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel. This (useless) example would not

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout