 Thread: RE: SOHO payment |
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HI,
Recently my subscription was extended with one year. Payment through my private Paypal account. I want to change this payment to my Business Paypal account. How to do this? In Paypal I can only choose for "cancel subscription", but that's not what I want.
Peter
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| Thread: RE: Unknown RTP codec 126 received > incoming calls dropped |
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After incoming calls are dropped, I noticed the line "Unknown RTP codec 126 received". When the SIP provider is configured directly to a phone (device or soft-phone), it is processed without errors. I cannot use the digital receptionist neither my answering service. Anyone's help is appreciated because nobody can reach me now!!!
Peter
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| Thread: No connection on incomming calls |
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When an incoming call is answered, I cannot hear the remote caller. It seems that the problem lies in the the trunk (I use Voipcheap for 6 years now)
- no issues when calling between extensions
- not even the digital receptionist is working
- on the remotephone, it looks like the caller is immediately placed on hold.
- outgoing calls are established without any issue using Voipcheap.
- if a SIP device is configured to use Voipcheap directly, it works without issues.
It happened suddenly without any changes in the configuration in PBSes or the clients. But even when I change the settings in PBXes, the problems are not solved.
Is there any know issue with Voipcheap lately? I'm lost, trying to figure this out for hours now but no luck.
Peter
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| Thread: RE: SIP calls are dropped ocasionally |
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Calls are dropped, no idea what caused it. New attempts are marked as "internal server error:
Mar 5 11:04:40 VERBOSE[103022] logger.c: -- Called [...]
Mar 5 11:04:40 VERBOSE[116155] logger.c: -- Got SIP response 500 "Internal server error"
Mar 5 11:04:40 VERBOSE[103022] logger.c: -- SIP/[...]-3cf3 is circuit-busy
Mar 5 11:04:40 VERBOSE[103022] chan_sip.c: Hangup call SIP/[...]-3cf3, SIP callid 6bfae8de53e8fedb0aa0cb647865582b@sip.voipcheap.com
I wonder if this is a problem locally, at PBXes or at my VOIP provider (voipcheap)?
Onyone?
Peter
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| Thread: RE: Intercom settings |
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Hmm, I've read these threads but it isn't clear to me what exactly should be changed. The author refers to an Asterisk server, which is (if I'm correct) a reference to PBXes. But these configurations cannot be changed by me (SoHo).
Maybe a moderator (i-p?) can be of assistance on this matter?
Peter
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| Thread: RE: Intercom settings |
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I can dial any extension and the phone rings and can be answered, but one of the features of an intercom is to have the call be anto answered (e.g. by a secretary or v.v.).
Everything that has to be setup for intercom is configured on the phone side: () only for explanation
sip intercom type: 2 (1=phone side | 2=server side) **
sip intercom prefix code: (??)
sip intercom line: 1
sip allow auto answer: 1 (1=true | 0=false)
sip intercom mute mic: 0 (1=true | 0=false)
sip intercom warning tone: 1 (1=true | 0=false)
sip intercom allow barge in: 1 (1=true | 0=false)
sip early media mute mic: 0 (1=true | 0=false)
** according to the admin manual of the phone: "This parameter is required for all server-side Intercom calls."
Hope you are able to help me with this.
Thx.
Peter
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| Thread: RE: Intercom settings |
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Hi there,
I have 5 Aastra 67571 SIP phones and I want to enable an intercom feature between 2 extensions. Everything is configured on the phone side, but I cannot get it to work. Is there a special prefix to use for PBXes for intercom calls (I've tried ** but that didn't work)?
Peter
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| Thread: Digital Receptionist dialing codes not recognized |
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I've created 2 digital receptionist menus with 2 options. The first menu with dialing code 1 or 2 is working fine. The second menu has also dialing code 1 and 2 but numbers pressed by the caller are not recognized. The dialing codes of both menus result to the same actions (transfer to extension voicemail) I've also tested to change the dialing codes to numbers 3 and 4, and connect the dialing codes to other actions. Nothing helps. I can't figure out what I'm doing wrong since the first menu is working as it should be. Or is this a bug in pbxes?
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