similar to: paging thru sipura-841

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other

We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats" on client machine shows more and more dropped packets on the

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

Dear All Greetings. I have a question regarding ext3 file system. I have installed Red Hat linux 8 with ext2 file system and I have multiple partition. Now I want to convert them to ext3 file system without distrubing my data. If any one got the idea, please let me know. Thanks Muhammad Noman -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

Hello, I've problems with following - ----- --- --- PSTN | --- isdn --- | A | ----- iax2 ------ | B | ----- --- --- On [B], there is unconditional call forwarding set back via [A] (dialparties.agi is used) to PSTN. So, call from PSTN is routed via [A] to [B] and than back again into PSTN.

Hi All - Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register => xxxx:xxxx@iaxtel.com bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 notransfer=yes context=incoming disallow=all ;

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

In my IAX config file I have: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 notransfer=yes allanrobertson- 209.23.224.97 (D) 255.255.255.255

Hi, I am new here on this list, and have a problem of which I hope that somebody here can help me with it. I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",

Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all

Is there any way to have the user confirm the extension they are looking to go to before transfering? i.e. "You pressed 5 4 3 3 2. Is this correct?" 1 - GoTo extensionPressed 2 - Enter extension again Thanks! Michael

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Surprisingly, Office B is the one loosing

I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13) and noticed a setting marked 'paging' under supplementary services on the Phone settings page on the advanced admin login. Anyone know how it might be used? Could it be like the Snom - exten => 10,1,SetVar(VXML_URL=intercom=true) exten => 10,2,Dial(SIP/testuser) Craig

I want to implement a one way announcement and paging facility using Asterisk and Sipura phones. The wiki says Sipura phones only support Auto Answer using the Call-Info header which is no lone shipped with asterisk stable since 1.0.4. I would like to ask if anyone has implemented a similiar facility using Sipura SPA-841 or any other SIP phones. If I could take a look at how