similar to: why do I get this message

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

Dear All, I have a tiny dial plan like: [testing] exten => 454,s,Ringing() exten => 454,n,Wait(4) exten => 454,n,Dial(SIP/slee@192.168.45.183:5605,10) exten => 454,n,Hangup This connects fine when I dial 454 from any extension in my system, but there is never any audio? Where can I start to look for debugging this? It's all internal so no NAT problems? Thanks, Gavin.

hi! First of all i want to tell i have a dedicated server on layeredtech with direct internet connection and i currently dont use iptables, so this is not about network configuration =). well so, i install asterisk-1.4.2 on my server, and next install asterisk-gui from the digium repository. next i go to: http://pbxa.com:8088/asterisk/static/config/cfgbasic.html and install a default

Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:wsiler@education2020.com <mailto:wsiler@education2020.com>

Hello eveybody, I've been looking for a way to share trunks between two asterisk servers. I guest I have to use Dundi, but I've not found the exact method yet. I need a way to allow users registered in one server to use the another server's trunks in the case the first server's trunks were busy and vice versa. Is this possible? Thank you so much, any comment will be useful.

Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =

I was just curious to what your redundancy solution is. I have considered many options, so I thought I would share and get an idea for what others are doing. My setup is two different locations with a 10MB WLAN fiber link between the two. Each location has it's own PRI as well. I have considered and tested many options this last year or so. 1) Using hearbeat and drbd to monitor the

Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and

I am not sure of the best way to do this, so I thought I would query the list. I have about 100 internal extensions ranging from 2000 - 2100. Each internal extension has a external DID number. For example: 2001 = 5552871620. As you can see the internal externsion and DID don't match in any way. What would be the best way to set the DID for when a extension dials out on the PRI? In

Hi....asterisk users... how to install asterisk on redhat ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070410/0e647e89/attachment-0001.htm

Hi List, I need to change my provider, at this time Asterisk box is on VOIP trunk. I have two options, T1 or 15 analog lines. I have some experience with analog and I have had two main issues with it. first is echo (I have not tried HPEC yet) and second unpredictable volume. The question is, if I use TE100 with PRI , will I have same issues? I would appreciate any comments and sample zaptel.conf

Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card & my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's "my account page" but still i have the same error when I attempt

All, I have done research on VoIP for some time now. I'm a Cisco certified Network Engineer however Telecom is not my strongest suit. I've been a part of this mailing list for sometime but my delusions of grandeur in migrating our 25 year old phone system to a new platform have been on the back burner, until now. I have found my company is moving to a new location(building) and this

I know over time SIP OPTIONS message handling has changed and I've seen some write ups that seem to indicate that an s extension in the default context is needed now to get them to work. It's probably my error in any case. So, what am I doing wrong or what do I need to do to get the sip ping to work? Bruce Ferrell Just for fun, I created a sip peer called ping at a fixed address

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

Does anyone know if is possible to purposely set red alarm status on PRI circuit for testing purposes (other than unplugging it). I have looked for a console command which might allow this.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070411/985e8e64/attachment.htm

Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

What wrong with this: [get-dnisinfo] ; sub-routine to get owner's password exten => s,1,Verbose( == ) exten => s,n,MYSQL(Connect connid localhost root password dax) exten => s,n,MYSQL(Query resultid ${connid} SELECT\ password\ FROM\ dnislookup\ WHERE\ dnis=\'${IVR-Exten}\') exten => s,n,MYSQL(Fetch fetchid ${password} password) exten => s,n,Verbose( == Password found

Can anyone tell me what the capacity is of 2 E1's in minutes. Ie how many minutes can 2 E1's take. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070413/de59fcf5/attachment.htm

Hi all, lets say I've registered at several Sip-Providers. Provider A offers best rates but is often too busy to get a line. Sip Provider B is stable (but more expensive). The asterisk box has a high call volume therefore problems at provider A will get obvious after a few calls stalled. In this case astersik shall switch temporarily to provider B but shall test periodically for selected