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ip pbx voip Multihoming and VOIP - Case[IAJG0902-8999H] (1 viewing) (1) Guests
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TOPIC: ip pbx voip Multihoming and VOIP - Case[IAJG0902-8999H]
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ip pbx voip Multihoming and VOIP - Case[IAJG0902-8999H]
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Hello Support, I recently added a second IP(on the same subnet) to my server in order to dedicate one IP for CGP. Since then the sip module has problems communicating from the server to the phone extensions. The box has only the 2 public IPs on one network card and all phones are connecting over the Internet and are behind NATed networks. The behavior is: I can't hear anything coming from the server to the phone(voicemails, autoatendant, voice conversations) but the far end party can her me, meaning voice is routed correctly from the phone to the server and out through the sip provider. Call from outside hear the autoatnedant, the recorded prompts and can leave voicemails so the communications from the serve to the SIP provider is fine, the communication from the server to the near end has problems in one direction phone-to-server. The settings are: Settings-General-Info shows both addresses as being used.I tried with only one IP and didn't help Network-NATed IPs has all the private networks enabled. Network-LAN IPs has WAN IPv4 address set to the second IP of the server, the one I intend to dedicate for CGP Real-Time-SIp-Receiving UDP and TCP have 5060 -second IP and 5060 -all addresses. So far I have been trying to add and remove the private IP ranges from Network-NATed IPs but then I add another problem where call from one domain cross over on another domain(to voice domains on this server) if I have 2 account set on one phone. A soft phone with only one account seemed to pass voice in both directions when NATed ranges have been disabled but not a Grandstream phone. The logs don't say anything out of ordinary or maybe I am overlooking that one important line but anyway I need some suggestions on where to look. Chris Vlad
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The administrator has disabled public write access. |
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ip pbx voip Multihoming and VOIP - Case[IAJG0902-8999H]
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Hello Support, I recently added a second IP(on the same subnet) to my server in order to dedicate one IP for CGP. Since then the sip module has problems communicating from the server to the phone extensions. The box has only the 2 public IPs on one network card and all phones are connecting over the Internet and are behind NATed networks. The behavior is: I can't hear anything coming from the server to the phone(voicemails, autoatendant, voice conversations) but the far end party can her me, meaning voice is routed correctly from the phone to the server and out through the sip provider. Call from outside hear the autoatnedant, the recorded prompts and can leave voicemails so the communications from the serve to the SIP provider is fine, the communication from the server to the near end has problems in one direction phone-to-server. The settings are: Settings-General-Info shows both addresses as being used.I tried with only one IP and didn't help Network-NATed IPs has all the private networks enabled. Network-LAN IPs has WAN IPv4 address set to the second IP of the server, the one I intend to dedicate for CGP Real-Time-SIp-Receiving UDP and TCP have 5060 -second IP and 5060 -all addresses. So far I have been trying to add and remove the private IP ranges from Network-NATed IPs but then I add another problem where call from one domain cross over on another domain(to voice domains on this server) if I have 2 account set on one phone. A soft phone with only one account seemed to pass voice in both directions when NATed ranges have been disabled but not a Grandstream phone. The logs don't say anything out of ordinary or maybe I am overlooking that one important line but anyway I need some suggestions on where to look. Chris Vlad
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The administrator has disabled public write access. |
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ip pbx voip Multihoming and VOIP - Case[IAJG0902-8999H]
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Hello Support, I recently added a second IP(on the same subnet) to my server in order to dedicate one IP for CGP. Since then the sip module has problems communicating from the server to the phone extensions. The box has only the 2 public IPs on one network card and all phones are connecting over the Internet and are behind NATed networks. The behavior is: I can't hear anything coming from the server to the phone(voicemails, autoatendant, voice conversations) but the far end party can her me, meaning voice is routed correctly from the phone to the server and out through the sip provider. Call from outside hear the autoatnedant, the recorded prompts and can leave voicemails so the communications from the serve to the SIP provider is fine, the communication from the server to the near end has problems in one direction phone-to-server. The settings are: Settings-General-Info shows both addresses as being used.I tried with only one IP and didn't help Network-NATed IPs has all the private networks enabled. Network-LAN IPs has WAN IPv4 address set to the second IP of the server, the one I intend to dedicate for CGP Real-Time-SIp-Receiving UDP and TCP have 5060 -second IP and 5060 -all addresses. So far I have been trying to add and remove the private IP ranges from Network-NATed IPs but then I add another problem where call from one domain cross over on another domain(to voice domains on this server) if I have 2 account set on one phone. A soft phone with only one account seemed to pass voice in both directions when NATed ranges have been disabled but not a Grandstream phone. The logs don't say anything out of ordinary or maybe I am overlooking that one important line but anyway I need some suggestions on where to look. Chris Vlad
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The administrator has disabled public write access. |
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ip pbx voip Multihoming and VOIP - Case[IAJG0902-8999H]
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Hello Support, I recently added a second IP(on the same subnet) to my server in order to dedicate one IP for CGP. Since then the sip module has problems communicating from the server to the phone extensions. The box has only the 2 public IPs on one network card and all phones are connecting over the Internet and are behind NATed networks. The behavior is: I can't hear anything coming from the server to the phone(voicemails, autoatendant, voice conversations) but the far end party can her me, meaning voice is routed correctly from the phone to the server and out through the sip provider. Call from outside hear the autoatnedant, the recorded prompts and can leave voicemails so the communications from the serve to the SIP provider is fine, the communication from the server to the near end has problems in one direction phone-to-server. The settings are: Settings-General-Info shows both addresses as being used.I tried with only one IP and didn't help Network-NATed IPs has all the private networks enabled. Network-LAN IPs has WAN IPv4 address set to the second IP of the server, the one I intend to dedicate for CGP Real-Time-SIp-Receiving UDP and TCP have 5060 -second IP and 5060 -all addresses. So far I have been trying to add and remove the private IP ranges from Network-NATed IPs but then I add another problem where call from one domain cross over on another domain(to voice domains on this server) if I have 2 account set on one phone. A soft phone with only one account seemed to pass voice in both directions when NATed ranges have been disabled but not a Grandstream phone. The logs don't say anything out of ordinary or maybe I am overlooking that one important line but anyway I need some suggestions on where to look. Chris Vlad
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The administrator has disabled public write access. |
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