similar to: no sound when bridging 2 asterisk SIP connections

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc.

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Hi, I have 2 linux accounts on different machines (same login, same password). Can you please tell me how I use rsync directories between 2 accounts? Thank you.

Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using

Hi, To push voice through a long thin wan (dsl) there are two choices: (1) have the cisco's (7912G) talk g729a to each other (reinvite=yes), or (2) have the cisco's talk to their local * in ulaw (reinvite=no), which talk to each other through a more advanced low-bandwidth codec (ilbc or speex) which is best? (2) would have more latency, wouldn't it? Did I miss a third option?

Hi: Can anybody point me to some guidelines about SMB network design or give some advice? Samba HOWTOs are very detailed recipes, but I need some general tips, like if we are serving fish or pasta tonight :-) This is the situation: a WAN with 20 offices with 2 to 30 people in each, plus a headquarter with 50 people, plus the databases and central file servers. The organization grew up on a NT4

Hello, We have several regional asterisk's connected to a central one making the the PRI calls through a TE410P card. When using SetCallingPres(prohibited) on a call at the regional level, that setting it not forwarded to the central asterisk and the call is made as if no callerid had been sent: the telco substitutes the network number. Using SetCallingPres(prohibited) on the central

Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful.

Hi, In our backup script we sometimes would like to override the common (i.e: static) excludes filter list. For example we exclude "- *.ext" for all backups but would like to include "+ *.ext" only for 'local' backups. Are such entries supposed to cancel each other? How can one override an earlier exclude in a filter list? Thanks,

Hi, I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by "zap show channels". I tried adding "dahdichanname = no" to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel

Hi, This used to work fine in 1.4: exten => 2131/,1,NoOp(reject3: ${CALLERID(num)}) exten => 2131/,n,Playback(no_unknow_callerid_here) exten => 2131/,n,Hangup And now, after upgrading to 1.6.1.x it matches every callerid. Did something change? Thanks,

Hi, Is there a way to have markdown automatically convert obvious (http, mailto) URL's to links? i.e: http://example.com -> <a href="http://example.com>http://example.com</a> Thanks, -- http://www.critikart.net

Hi, If several phones register to the same sip.conf section what will happen with a "Dial SIP/shared" in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? Thanks, -- Jesus is coming! Everyone look busy!

Hi out there, I`ve not a real problem (currently), but a few questions on wan-wide (read: low bandwidth) share distribution with samba 2.2.1a. The setup described below was developed in a hurry (was needed by tomorrow :-\). Are there better ways to do this? What are the pros and cons of using the msdfs - options in this situation? Has anyone set up a similar share topology? If so, would you

Hello everyone, this is my first post to the list, so hello again. We're a small company in Romania and we're trying to set up a really small version of "call center". That is, we want to get a few land-lines from our telco in different countys and "bridge" all calls to our HQ, in order to make it cheeper for our clients to call us. Unfortunatelly there's no ISP

Hello, Our company is trying to implement central Windows Domain at HQ and replicated across all its regional offices. The implementation will have a PDC/LDAP-master on HQ and BDC/LDAP-slave on each regional office. In the hopes of saving bandwidth we are trying to avoid the use of WINS between WAN links. Is there any way of not using WINS and still have the clients find the PDC (for updating

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

If like me you run * on a VPN (or multihomed) gateway and want to serve remote SIP clients, make sure you have bindaddr = 192.168.0.1 ; or whatever is your box's private IP otherwise * might bind to its public IP and send it as return address in the SIP call setup, which will (should) be rejected by your firewall. To * experts: might this setting interfer with NATed SIP clients? -- I

---------- Original Message ---------------------------------- From: Louis-David Mitterrand <vindex@apartia.org> Reply-To: asterisk-users@lists.digium.com Date: Mon, 22 Sep 2003 22:28:40 +0200 >On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote: >> My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of >> service. They were deployed for about 6