Re: [asterisk-users] Caller ID Names

2015-03-20 Thread Jordan Cook - Gyron Networks
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: 11 March 2015 17:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Names Are the

Re: [asterisk-users] Caller ID Names

2015-03-20 Thread Jordan Cook - Gyron Networks
From a softphone (x-lite) the caller id information comes through as anonymous@anonymous.invalid These are also valid calls - If I disable outbound CLID on my mobile and call in - this happens. However it works fine on calls where I send caller id information. Okay, just figured this

[asterisk-users] Caller ID Names

2015-03-10 Thread Jordan Cook - Gyron Networks
Hi, In my dialplan I have the following line. same = n,Set(CALLERID(name)=Support) I am expecting this to always set the caller id name to 'Support' - however, we are getting calls come in as Anonymous with the number as something like unknown@unknown We're using Cisco 7945 phones - I

[asterisk-users] Events

2015-03-02 Thread Jordan Cook - Gyron Networks
Hello, I am playing around with events in asterisk via asterisk manager - i've noticed it doesnt seem to be emitting events to my connected client. Is there something that I need to do to receive events? Also output from 'manager show events' voip*CLI manager show events Events:

Re: [asterisk-users] Polycom SoundStation 6000 Dropping Registration

2015-01-23 Thread Jordan Cook - Gyron Networks
We run a variety of 5000, 6000, and 7000 series Soundstations running Asterisk 11.6.0 and the phones are at 4.0.3.7562. We do not see these registration issues. Would you be willing to send the configuration from asterisk for this? This message may be private and confidential. If you have

[asterisk-users] Polycom SoundStation 6000 Dropping Registration

2015-01-23 Thread Jordan Cook - Gyron Networks
Hello, I'm having a problem with a few Polycom SoundStation 6000s. Everything works fine, but they drop registration to asterisk after about maybe 30 minutes - the phone does not re-try to register and if you try to dial out on the phone it says URI Dialing is Disabled Has anyone else had

Re: [asterisk-users] Problem with Cisco Phones

2015-01-22 Thread Jordan Cook - Gyron Networks
Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- version-9/ I've done some more playing about and what I've noticed is that even when using TCP SIP on the 8.x Firmware conferencing doesn’t work - making it use UDP

Re: [asterisk-users] Problem with Cisco Phones

2015-01-20 Thread Jordan Cook - Gyron Networks
Next step is packet capture to see if there is a clue as to the cause of the failure in the SIP signalling. Right, I see the following when running SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? --- SIP

[asterisk-users] Problem with Cisco Phones

2015-01-20 Thread Jordan Cook - Gyron Networks
Possibly slightly off topic, has anyone ever had Cisco 79xx Series phones come up with cannot complete conference errors when trying to conference two calls together? This message may be private and confidential. If you have received this message in error, please notify us and remove it from

Re: [asterisk-users] Problem with Cisco Phones

2015-01-20 Thread Jordan Cook - Gyron Networks
We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred, any other ideas? I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail.