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- About Saja Ranjbari
04-11-2019
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02-23-2018
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Created by
Saja Ranjbari
in IP Telephony and Phones
in IP Telephony and Phones
03-28-2018
03:33 AM
03-28-2018
03:33 AM
Could you resolve your issue ?? becuase i have this issue too
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Created by
Saja Ranjbari
in IP Telephony and Phones
in IP Telephony and Phones
03-27-2018
11:43 PM
03-27-2018
11:43 PM
Hello All
We have this issue too with softx3000,
any calls (incoming and outgoing) cant work. becuase technicals of provider said "we cant recive any response by your GW" but router 2811 are sending 180 , and after a while i give 504 error gateway time our error with this trying
Provider SBC
Huawei SoftX3000 Cisco CME IP Phone
-------------------------------->Router 2811<------------ ^ extension 105
i attached trace file and picture of wireshar,
is there anyone that has experianceabout this issue ?
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Created by
Saja Ranjbari
in IP Telephony and Phones
in IP Telephony and Phones
03-27-2018
12:37 AM
03-27-2018
12:37 AM
10.123.101.165 is SBC of Provider 10.194.22.222 our IP PBX
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Created by
Saja Ranjbari
in IP Telephony and Phones
in IP Telephony and Phones
03-24-2018
08:41 AM
03-24-2018
08:41 AM
Yes sure,
PFA
tnq
Nipun Singh Ragha
(just incoming works, outgoing can't work), with SBC softx3000 Huawei
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Created by
Saja Ranjbari
in IP Telephony and Phones
in IP Telephony and Phones
03-24-2018
08:02 AM
03-24-2018
08:02 AM
just incoming can work but for outgoing it doesn't work , because it choose another port,
and it's results of it.
#Show sip-ua calls
15.X (can't work)
Call 1 SIP Call ID : isbcihmxuuwusvcj33mvgcnxqt5uw5isijxu@SoftX3000 State of the call : STATE_SENT_ALERTING (14) Substate of the call : SUBSTATE_NONE (0) Calling Number : 9378378583 Called Number : 49085 Bit Flags : 0x8C4001C 0x10000140 0x444 CC Call ID : 78 Source IP Address (Sig ): 10.194.22.200 Destn SIP Req Addr:Port : [10.123.101.100]:5060 Destn SIP Resp Addr:Port: [10.123.101.100]:5060 Destination Name : 10.123.101.100 Number of Media Streams : 1 Number of Active Streams: 0 RTP Fork Object : 0x0 Media Mode : flow-through Media Stream 1 State of the stream : STREAM_ADDING Stream Call ID : 78 Stream Type : voice-only (0) Stream Media Addr Type : 1 Negotiated Codec : g711alaw (160 bytes) Codec Payload Type : 8 Negotiated Dtmf-relay : inband-voice Dtmf-relay Payload Type : 0 QoS ID : -1 Local QoS Strength : BestEffort Negotiated QoS Strength : BestEffort Negotiated QoS Direction : None Local QoS Status : None Media Source IP Addr:Port: [10.194.22.200]:18384 Media Dest IP Addr:Port : [10.123.101.100]:20002 ****************************************************************** 12.4 (just incoming works) Call 1 SIP Call ID : isbc55xs3tig5nixi35hiwnv5mnjuvvij5xn@SoftX3000 State of the call : STATE_DEAD (10) Substate of the call : SUBSTATE_NONE (0) Calling Number : 9378378583 Called Number : 49085 Bit Flags : 0x884008C 0x300 0x800004 CC Call ID : 11 Source IP Address (Sig ): 10.194.22.200 Destn SIP Req Addr:Port : [10.123.101.100]:5060 Destn SIP Resp Addr:Port: [10.123.101.100]:5060 Destination Name : 10.123.101.100 Number of Media Streams : 1 Number of Active Streams: 0 RTP Fork Object : 0x0 Media Mode : flow-through Media Stream 1 State of the stream : STREAM_DEAD Stream Call ID : 11 Stream Type : voice-only (0) Stream Media Addr Type : 1 Negotiated Codec : g711alaw (160 bytes) Codec Payload Type : 8 Negotiated Dtmf-relay : inband-voice Dtmf-relay Payload Type : 0 QoS ID : -1 Local QoS Strength : BestEffort Negotiated QoS Strength : BestEffort Negotiated QoS Direction : None Local QoS Status : None Media Source IP Addr:Port: [10.194.22.200]:16858 Media Dest IP Addr:Port : [10.123.101.100]:48458
and please check this picture in this picture after choosing the codec i think for choise RTP port it can't be success and occur error 500
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Created by
Saja Ranjbari
in IP Telephony and Phones
in IP Telephony and Phones
03-24-2018
06:57 AM
03-24-2018
06:57 AM
I checked our sip trunk link with Asterisk, Asterisk isn't other switch, it was just for test link, in fact, I have one CME Cisco (2811) in my border network, and one sip trunk link from sip provider like callcenteric, I have to change my setting to best setting, like change RTP port to 16000. all of the outgoing and incoming calls come from this sip trunk link.
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Created by
Saja Ranjbari
in IP Telephony and Phones
in IP Telephony and Phones
03-23-2018
11:52 PM
03-23-2018
11:52 PM
I have some issue with incoming calls and outgoing calls with sip trunk provider, all of the processes for initial session sip trunk can work but to choose RTP port it had issue (no voice, no ringback tone in my cell phone, and it show dialing, same time in router calls come), I checked it with asterisk family and it can work very good but for Cisco, it doesn't work good, I checked many commands like MTP and transcoder and NTE and min-sec etc,
in final with many trying I found it (with change ios when RTP port start with 16000 every call can work but for outgoing calls it changes RTP port to 18000 and can't work again, in another (most of it) version ios of cisco RTP port starts from 18000 and sometimes 19000, and issue again occurs
I am trying to find some command that with those can choose and specific RTP port range for CME, i found some command but it can't work on CME just works on CUBE
i found this command for change RTP range, BUT
Device(config)# voice service voip
Device(conf-voi-serv)# allow-connections sip to sip
Device(config-voi-serv)# media-address range 2001:DB8::/48
Device(config-voi-serv)# rtp-port range 20000 30000
But it doesn't work on CME ISR G1 like (2811), and in this issue,
is there anyone that can give any suggestion or command that can help??
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Labels:
03-23-2018
02:53 PM
Many Thanks for share this link, but this information is about CUBE
i found this command for change RTP range ,BUT
Device(config)# voice service voip
Device(conf-voi-serv)# allow-connections sip to sip
Device(config-voi-serv)# media-address range 2001:DB8::/48
Device(config-voi-serv)# rtp-port range 20000 30000
But it doesn't work on CME ISR G1 like (2811), and i have this issue too,
is there anyone that can give any suggestion or command that can help us ??
... View more
Created by
Saja Ranjbari
in IP Telephony and Phones
03-28-2018
03-28-2018
Could you resolve your issue ?? becuase i have this issue too
03-27-2018
Hello All We have this issue too with softx3000, any calls (incoming and
outgoing) cant work. becuas...
03-24-2018
Yes sure, PFA tnq Nipun Singh Ragha (just incoming works, outgoing can't
work), with SBC softx3000 H...
03-24-2018
just incoming can work but for outgoing it doesn't work , because it
choose another port, and it's r...
03-24-2018
I checked our sip trunk link with Asterisk, Asterisk isn't other switch,
it was just for test link,i...
03-23-2018
I have some issue with incoming calls and outgoing calls with sip trunk
provider, all of the process...
03-23-2018
Many Thanks for share this link, but this information is about CUBE i
found this command for change ...
Created by
Saja Ranjbari
in Unified Communications Infrastructure
02-23-2018
02-23-2018
Hi Jaime Thank you for your good information. I have one question. for
creating new order if we want...
Public Statistics
| Date Registered | 02-23-2018 12:19 PM |
| Date Last Visited | 04-11-2019 09:53 AM |
| Total Messages Posted | 11 |
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|---|---|
| 5 | |
| 5 | |
| 5 | |
| 5 | |
| 5 |
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