Входящие звонки уходят не неверный peer

Добрый день ! Asterisk чудит. Номера звонок уходит через неправильный peer на неправильный context Желаемый peer - 961538 Желаемый context - sklad Пользуемся услугами провайдера, который предоставляет нам 3 номера по sip. При подключении 3-го номера появилась проблема. Входящий звонок перехватывает другой peer. user'ы одинаковые за исключением username, password,context. Желаемый получатель 961538, звонок уходит на 948909. Прикладываю debug:

<------------> [Oct 3 10:44:35] VERBOSE[24436] chan_sip.c:

.<--- SIP read from UDP:IP SIP PROVIDER:5060 --->

.INVITE sip:961538@IP ASTEISK:5060 SIP/2.0

.Via: SIP/2.0/UDP IP SIP PROVIDER:5060;branch=z9hG4bK-o26d68dK2288418712926

.From: "9116481516" <sip:9116481516@ip sip="" provider;user="phone">;tag=26d68dK2288418712868

.To: <sip:961538@ip asteisk;user="phone">

.Call-ID: 1412-319027-290586

.CSeq: 290078 INVITE

.User-Agent: Eltex SMG SIP 2.12.24

.Contact: <sip:9116481516@ip sip="" provider:5060="">

.Accept: multipart/mixed, application/sdp

.Allow: INVITE, ACK, BYE, CANCEL, PRACK, REGISTER, INFO, REFER, NOTIFY, OPTIONS

.Supported: replaces

.Category: 10

.P-Eltex-Info: - {user,376} 1534 <0.12270.516>

.Content-Type: application/sdp

.Content-Length: 214

.v=0

.o=- 1534 1534 IN IP4 IP SIP PROVIDER

.s=SMG SIP session

.c=IN IP4 IP SIP PROVIDER

.t=0 0

.m=audio 53560 RTP/AVP 0 8 18

.a=rtpmap:0 PCMU/8000

.a=rtpmap:8 PCMA/8000

.a=rtpmap:18 G729/8000

.a=ptime:30

.a=sendrecv

.<-------------> [Oct 3 10:44:35] VERBOSE[24436] chan_sip.c: --- (15 headers 11 lines) ---

[Oct 3 10:44:35] VERBOSE[24436] chan_sip.c: Sending to IP SIP PROVIDER:5060 (no NAT)

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Sending to IP SIP PROVIDER:5060 (no NAT)

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Using INVITE request as basis request - 1412-319027-290586

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Found peer '948909' for '9116481516' from IP SIP PROVIDER:5060

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Found RTP audio format 0

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Found RTP audio format 8

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Found RTP audio format 18

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Found audio description format G729 for ID 18

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c: Peer audio RTP is at port IP SIP PROVIDER:53560

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chansip.c: Looking for 961538 in DID948909 (domain IP ASTEISK)

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chansip.c: listroute: hop: <sip:9116481516@ip sip="" provider:5060="">

[Oct 3 10:44:35] VERBOSE[24436][C-00000000] chan_sip.c:

.<--- Transmitting (no NAT) to IP SIP PROVIDER:5060 --->

.SIP/2.0 100 Trying

.Via: SIP/2.0/UDP IP SIP PROVIDER:5060;branch=z9hG4bK-o26d68dK2288418712926;received=IP SIP PROVIDER

.From: "9116481516" <sip:9116481516@ip sip="" provider;user="phone">;tag=26d68dK2288418712868

.To: <sip:961538@ip asteisk;user="phone">

.Call-ID: 1412-319027-290586

.CSeq: 290078 INVITE

.Server: Asterisk PBX

.Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

.Supported: replaces, timer

.Contact: <sip:961538@ip asteisk:5060="">

.Content-Length: 0

.<------------> [Oct 3 10:44:35] VERBOSE[24454][C-00000000] pbx.c: -- Executing [961538@DID_948909:1] Set("SIP/948909-00000000", "CALLERID(all)=+79116481516") in new stack

[Oct 3 10:44:35] VERBOSE[24454][C-00000000] pbx.c: -- Executing [961538@DID948909:2] Goto("SIP/948909-00000000", "companyinfo,s,1") in new stack

[Oct 3 10:44:35] VERBOSE[24454][C-00000000] pbx.c: -- Goto (company_info,s,1)

[Oct 3 10:44:35] VERBOSE[24454][C-00000000] pbx.c: -- Executing [s@company_info:1] Answer("SIP/948909-00000000", "") in new stack

[Oct 3 10:44:35] VERBOSE[24454][C-00000000] chan_sip.c: Audio is at 15756

[Oct 3 10:44:35] VERBOSE[24454][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP

[Oct 3 10:44:35] VERBOSE[24454][C-00000000] chan_sip.c: Adding codec 100004 (alaw) to SDP

[Oct 3 10:44:35] VERBOSE[24454][C-00000000] chan_sip.c:

.<--- Reliably Transmitting (no NAT) to IP SIP PROVIDER:5060 --->

.SIP/2.0 200 OK

.Via: SIP/2.0/UDP IP SIP PROVIDER:5060;branch=z9hG4bK-o26d68dK2288418712926;received=IP SIP PROVIDER

.From: "9116481516" <sip:9116481516@ip sip="" provider;user="phone">;tag=26d68dK2288418712868

.To: <sip:961538@ip asteisk;user="phone">;tag=as7443c85b

.Call-ID: 1412-319027-290586

.CSeq: 290078 INVITE

.Server: Asterisk PBX

.Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

.Supported: replaces, timer

.Contact: <sip:961538@ip asteisk:5060="">

.Content-Type: application/sdp

.Content-Length: 206